Threads.cpp revision 7844f679be8d94c5cdf017f53754cb68ee2f00da
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 139// So for now we just assume that client is double-buffered for fast tracks. 140// FIXME It would be better for client to tell AudioFlinger the value of N, 141// so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 2; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title 189#ifndef DEBUG_CPU_USAGE 190 __unused 191#endif 192 ) { 193#ifdef DEBUG_CPU_USAGE 194 // get current thread's delta CPU time in wall clock ns 195 double wcNs; 196 bool valid = mCpuUsage.sampleAndEnable(wcNs); 197 198 // record sample for wall clock statistics 199 if (valid) { 200 mWcStats.sample(wcNs); 201 } 202 203 // get the current CPU number 204 int cpuNum = sched_getcpu(); 205 206 // get the current CPU frequency in kHz 207 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 208 209 // check if either CPU number or frequency changed 210 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 211 mCpuNum = cpuNum; 212 mCpukHz = cpukHz; 213 // ignore sample for purposes of cycles 214 valid = false; 215 } 216 217 // if no change in CPU number or frequency, then record sample for cycle statistics 218 if (valid && mCpukHz > 0) { 219 double cycles = wcNs * cpukHz * 0.000001; 220 mHzStats.sample(cycles); 221 } 222 223 unsigned n = mWcStats.n(); 224 // mCpuUsage.elapsed() is expensive, so don't call it every loop 225 if ((n & 127) == 1) { 226 long long elapsed = mCpuUsage.elapsed(); 227 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 228 double perLoop = elapsed / (double) n; 229 double perLoop100 = perLoop * 0.01; 230 double perLoop1k = perLoop * 0.001; 231 double mean = mWcStats.mean(); 232 double stddev = mWcStats.stddev(); 233 double minimum = mWcStats.minimum(); 234 double maximum = mWcStats.maximum(); 235 double meanCycles = mHzStats.mean(); 236 double stddevCycles = mHzStats.stddev(); 237 double minCycles = mHzStats.minimum(); 238 double maxCycles = mHzStats.maximum(); 239 mCpuUsage.resetElapsed(); 240 mWcStats.reset(); 241 mHzStats.reset(); 242 ALOGD("CPU usage for %s over past %.1f secs\n" 243 " (%u mixer loops at %.1f mean ms per loop):\n" 244 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 245 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 246 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 247 title.string(), 248 elapsed * .000000001, n, perLoop * .000001, 249 mean * .001, 250 stddev * .001, 251 minimum * .001, 252 maximum * .001, 253 mean / perLoop100, 254 stddev / perLoop100, 255 minimum / perLoop100, 256 maximum / perLoop100, 257 meanCycles / perLoop1k, 258 stddevCycles / perLoop1k, 259 minCycles / perLoop1k, 260 maxCycles / perLoop1k); 261 262 } 263 } 264#endif 265}; 266 267// ---------------------------------------------------------------------------- 268// ThreadBase 269// ---------------------------------------------------------------------------- 270 271AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 272 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 273 : Thread(false /*canCallJava*/), 274 mType(type), 275 mAudioFlinger(audioFlinger), 276 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 277 // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 278 mParamStatus(NO_ERROR), 279 //FIXME: mStandby should be true here. Is this some kind of hack? 280 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 281 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 282 // mName will be set by concrete (non-virtual) subclass 283 mDeathRecipient(new PMDeathRecipient(this)) 284{ 285} 286 287AudioFlinger::ThreadBase::~ThreadBase() 288{ 289 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 290 for (size_t i = 0; i < mConfigEvents.size(); i++) { 291 delete mConfigEvents[i]; 292 } 293 mConfigEvents.clear(); 294 295 mParamCond.broadcast(); 296 // do not lock the mutex in destructor 297 releaseWakeLock_l(); 298 if (mPowerManager != 0) { 299 sp<IBinder> binder = mPowerManager->asBinder(); 300 binder->unlinkToDeath(mDeathRecipient); 301 } 302} 303 304status_t AudioFlinger::ThreadBase::readyToRun() 305{ 306 status_t status = initCheck(); 307 if (status == NO_ERROR) { 308 ALOGI("AudioFlinger's thread %p ready to run", this); 309 } else { 310 ALOGE("No working audio driver found."); 311 } 312 return status; 313} 314 315void AudioFlinger::ThreadBase::exit() 316{ 317 ALOGV("ThreadBase::exit"); 318 // do any cleanup required for exit to succeed 319 preExit(); 320 { 321 // This lock prevents the following race in thread (uniprocessor for illustration): 322 // if (!exitPending()) { 323 // // context switch from here to exit() 324 // // exit() calls requestExit(), what exitPending() observes 325 // // exit() calls signal(), which is dropped since no waiters 326 // // context switch back from exit() to here 327 // mWaitWorkCV.wait(...); 328 // // now thread is hung 329 // } 330 AutoMutex lock(mLock); 331 requestExit(); 332 mWaitWorkCV.broadcast(); 333 } 334 // When Thread::requestExitAndWait is made virtual and this method is renamed to 335 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 336 requestExitAndWait(); 337} 338 339status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 340{ 341 status_t status; 342 343 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 344 Mutex::Autolock _l(mLock); 345 346 mNewParameters.add(keyValuePairs); 347 mWaitWorkCV.signal(); 348 // wait condition with timeout in case the thread loop has exited 349 // before the request could be processed 350 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 351 status = mParamStatus; 352 mWaitWorkCV.signal(); 353 } else { 354 status = TIMED_OUT; 355 } 356 return status; 357} 358 359void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 360{ 361 Mutex::Autolock _l(mLock); 362 sendIoConfigEvent_l(event, param); 363} 364 365// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 366void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 367{ 368 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 369 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 370 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 371 param); 372 mWaitWorkCV.signal(); 373} 374 375// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 376void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 377{ 378 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 379 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 380 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 381 mConfigEvents.size(), pid, tid, prio); 382 mWaitWorkCV.signal(); 383} 384 385void AudioFlinger::ThreadBase::processConfigEvents() 386{ 387 Mutex::Autolock _l(mLock); 388 processConfigEvents_l(); 389} 390 391// post condition: mConfigEvents.isEmpty() 392void AudioFlinger::ThreadBase::processConfigEvents_l() 393{ 394 while (!mConfigEvents.isEmpty()) { 395 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 396 ConfigEvent *event = mConfigEvents[0]; 397 mConfigEvents.removeAt(0); 398 // release mLock before locking AudioFlinger mLock: lock order is always 399 // AudioFlinger then ThreadBase to avoid cross deadlock 400 mLock.unlock(); 401 switch (event->type()) { 402 case CFG_EVENT_PRIO: { 403 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 404 // FIXME Need to understand why this has be done asynchronously 405 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 406 true /*asynchronous*/); 407 if (err != 0) { 408 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 409 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 410 } 411 } break; 412 case CFG_EVENT_IO: { 413 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 414 { 415 Mutex::Autolock _l(mAudioFlinger->mLock); 416 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 417 } 418 } break; 419 default: 420 ALOGE("processConfigEvents() unknown event type %d", event->type()); 421 break; 422 } 423 delete event; 424 mLock.lock(); 425 } 426} 427 428void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 429{ 430 const size_t SIZE = 256; 431 char buffer[SIZE]; 432 String8 result; 433 434 bool locked = AudioFlinger::dumpTryLock(mLock); 435 if (!locked) { 436 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 437 write(fd, buffer, strlen(buffer)); 438 } 439 440 snprintf(buffer, SIZE, "io handle: %d\n", mId); 441 result.append(buffer); 442 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 443 result.append(buffer); 444 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 445 result.append(buffer); 446 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 447 result.append(buffer); 448 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 449 result.append(buffer); 450 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 451 result.append(buffer); 452 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 453 result.append(buffer); 454 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 455 result.append(buffer); 456 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 457 result.append(buffer); 458 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 459 result.append(buffer); 460 461 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 462 result.append(buffer); 463 result.append(" Index Command"); 464 for (size_t i = 0; i < mNewParameters.size(); ++i) { 465 snprintf(buffer, SIZE, "\n %02d ", i); 466 result.append(buffer); 467 result.append(mNewParameters[i]); 468 } 469 470 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 471 result.append(buffer); 472 for (size_t i = 0; i < mConfigEvents.size(); i++) { 473 mConfigEvents[i]->dump(buffer, SIZE); 474 result.append(buffer); 475 } 476 result.append("\n"); 477 478 write(fd, result.string(), result.size()); 479 480 if (locked) { 481 mLock.unlock(); 482 } 483} 484 485void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 486{ 487 const size_t SIZE = 256; 488 char buffer[SIZE]; 489 String8 result; 490 491 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 492 write(fd, buffer, strlen(buffer)); 493 494 for (size_t i = 0; i < mEffectChains.size(); ++i) { 495 sp<EffectChain> chain = mEffectChains[i]; 496 if (chain != 0) { 497 chain->dump(fd, args); 498 } 499 } 500} 501 502void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 503{ 504 Mutex::Autolock _l(mLock); 505 acquireWakeLock_l(uid); 506} 507 508String16 AudioFlinger::ThreadBase::getWakeLockTag() 509{ 510 switch (mType) { 511 case MIXER: 512 return String16("AudioMix"); 513 case DIRECT: 514 return String16("AudioDirectOut"); 515 case DUPLICATING: 516 return String16("AudioDup"); 517 case RECORD: 518 return String16("AudioIn"); 519 case OFFLOAD: 520 return String16("AudioOffload"); 521 default: 522 ALOG_ASSERT(false); 523 return String16("AudioUnknown"); 524 } 525} 526 527void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 528{ 529 getPowerManager_l(); 530 if (mPowerManager != 0) { 531 sp<IBinder> binder = new BBinder(); 532 status_t status; 533 if (uid >= 0) { 534 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 535 binder, 536 getWakeLockTag(), 537 String16("media"), 538 uid); 539 } else { 540 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 541 binder, 542 getWakeLockTag(), 543 String16("media")); 544 } 545 if (status == NO_ERROR) { 546 mWakeLockToken = binder; 547 } 548 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 549 } 550} 551 552void AudioFlinger::ThreadBase::releaseWakeLock() 553{ 554 Mutex::Autolock _l(mLock); 555 releaseWakeLock_l(); 556} 557 558void AudioFlinger::ThreadBase::releaseWakeLock_l() 559{ 560 if (mWakeLockToken != 0) { 561 ALOGV("releaseWakeLock_l() %s", mName); 562 if (mPowerManager != 0) { 563 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 564 } 565 mWakeLockToken.clear(); 566 } 567} 568 569void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 570 Mutex::Autolock _l(mLock); 571 updateWakeLockUids_l(uids); 572} 573 574void AudioFlinger::ThreadBase::getPowerManager_l() { 575 576 if (mPowerManager == 0) { 577 // use checkService() to avoid blocking if power service is not up yet 578 sp<IBinder> binder = 579 defaultServiceManager()->checkService(String16("power")); 580 if (binder == 0) { 581 ALOGW("Thread %s cannot connect to the power manager service", mName); 582 } else { 583 mPowerManager = interface_cast<IPowerManager>(binder); 584 binder->linkToDeath(mDeathRecipient); 585 } 586 } 587} 588 589void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 590 591 getPowerManager_l(); 592 if (mWakeLockToken == NULL) { 593 ALOGE("no wake lock to update!"); 594 return; 595 } 596 if (mPowerManager != 0) { 597 sp<IBinder> binder = new BBinder(); 598 status_t status; 599 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 600 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 601 } 602} 603 604void AudioFlinger::ThreadBase::clearPowerManager() 605{ 606 Mutex::Autolock _l(mLock); 607 releaseWakeLock_l(); 608 mPowerManager.clear(); 609} 610 611void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 612{ 613 sp<ThreadBase> thread = mThread.promote(); 614 if (thread != 0) { 615 thread->clearPowerManager(); 616 } 617 ALOGW("power manager service died !!!"); 618} 619 620void AudioFlinger::ThreadBase::setEffectSuspended( 621 const effect_uuid_t *type, bool suspend, int sessionId) 622{ 623 Mutex::Autolock _l(mLock); 624 setEffectSuspended_l(type, suspend, sessionId); 625} 626 627void AudioFlinger::ThreadBase::setEffectSuspended_l( 628 const effect_uuid_t *type, bool suspend, int sessionId) 629{ 630 sp<EffectChain> chain = getEffectChain_l(sessionId); 631 if (chain != 0) { 632 if (type != NULL) { 633 chain->setEffectSuspended_l(type, suspend); 634 } else { 635 chain->setEffectSuspendedAll_l(suspend); 636 } 637 } 638 639 updateSuspendedSessions_l(type, suspend, sessionId); 640} 641 642void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 643{ 644 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 645 if (index < 0) { 646 return; 647 } 648 649 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 650 mSuspendedSessions.valueAt(index); 651 652 for (size_t i = 0; i < sessionEffects.size(); i++) { 653 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 654 for (int j = 0; j < desc->mRefCount; j++) { 655 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 656 chain->setEffectSuspendedAll_l(true); 657 } else { 658 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 659 desc->mType.timeLow); 660 chain->setEffectSuspended_l(&desc->mType, true); 661 } 662 } 663 } 664} 665 666void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 667 bool suspend, 668 int sessionId) 669{ 670 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 671 672 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 673 674 if (suspend) { 675 if (index >= 0) { 676 sessionEffects = mSuspendedSessions.valueAt(index); 677 } else { 678 mSuspendedSessions.add(sessionId, sessionEffects); 679 } 680 } else { 681 if (index < 0) { 682 return; 683 } 684 sessionEffects = mSuspendedSessions.valueAt(index); 685 } 686 687 688 int key = EffectChain::kKeyForSuspendAll; 689 if (type != NULL) { 690 key = type->timeLow; 691 } 692 index = sessionEffects.indexOfKey(key); 693 694 sp<SuspendedSessionDesc> desc; 695 if (suspend) { 696 if (index >= 0) { 697 desc = sessionEffects.valueAt(index); 698 } else { 699 desc = new SuspendedSessionDesc(); 700 if (type != NULL) { 701 desc->mType = *type; 702 } 703 sessionEffects.add(key, desc); 704 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 705 } 706 desc->mRefCount++; 707 } else { 708 if (index < 0) { 709 return; 710 } 711 desc = sessionEffects.valueAt(index); 712 if (--desc->mRefCount == 0) { 713 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 714 sessionEffects.removeItemsAt(index); 715 if (sessionEffects.isEmpty()) { 716 ALOGV("updateSuspendedSessions_l() restore removing session %d", 717 sessionId); 718 mSuspendedSessions.removeItem(sessionId); 719 } 720 } 721 } 722 if (!sessionEffects.isEmpty()) { 723 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 724 } 725} 726 727void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 728 bool enabled, 729 int sessionId) 730{ 731 Mutex::Autolock _l(mLock); 732 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 733} 734 735void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 736 bool enabled, 737 int sessionId) 738{ 739 if (mType != RECORD) { 740 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 741 // another session. This gives the priority to well behaved effect control panels 742 // and applications not using global effects. 743 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 744 // global effects 745 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 746 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 747 } 748 } 749 750 sp<EffectChain> chain = getEffectChain_l(sessionId); 751 if (chain != 0) { 752 chain->checkSuspendOnEffectEnabled(effect, enabled); 753 } 754} 755 756// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 757sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 758 const sp<AudioFlinger::Client>& client, 759 const sp<IEffectClient>& effectClient, 760 int32_t priority, 761 int sessionId, 762 effect_descriptor_t *desc, 763 int *enabled, 764 status_t *status) 765{ 766 sp<EffectModule> effect; 767 sp<EffectHandle> handle; 768 status_t lStatus; 769 sp<EffectChain> chain; 770 bool chainCreated = false; 771 bool effectCreated = false; 772 bool effectRegistered = false; 773 774 lStatus = initCheck(); 775 if (lStatus != NO_ERROR) { 776 ALOGW("createEffect_l() Audio driver not initialized."); 777 goto Exit; 778 } 779 780 // Allow global effects only on offloaded and mixer threads 781 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 782 switch (mType) { 783 case MIXER: 784 case OFFLOAD: 785 break; 786 case DIRECT: 787 case DUPLICATING: 788 case RECORD: 789 default: 790 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 791 lStatus = BAD_VALUE; 792 goto Exit; 793 } 794 } 795 796 // Only Pre processor effects are allowed on input threads and only on input threads 797 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 798 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 799 desc->name, desc->flags, mType); 800 lStatus = BAD_VALUE; 801 goto Exit; 802 } 803 804 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 805 806 { // scope for mLock 807 Mutex::Autolock _l(mLock); 808 809 // check for existing effect chain with the requested audio session 810 chain = getEffectChain_l(sessionId); 811 if (chain == 0) { 812 // create a new chain for this session 813 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 814 chain = new EffectChain(this, sessionId); 815 addEffectChain_l(chain); 816 chain->setStrategy(getStrategyForSession_l(sessionId)); 817 chainCreated = true; 818 } else { 819 effect = chain->getEffectFromDesc_l(desc); 820 } 821 822 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 823 824 if (effect == 0) { 825 int id = mAudioFlinger->nextUniqueId(); 826 // Check CPU and memory usage 827 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 828 if (lStatus != NO_ERROR) { 829 goto Exit; 830 } 831 effectRegistered = true; 832 // create a new effect module if none present in the chain 833 effect = new EffectModule(this, chain, desc, id, sessionId); 834 lStatus = effect->status(); 835 if (lStatus != NO_ERROR) { 836 goto Exit; 837 } 838 effect->setOffloaded(mType == OFFLOAD, mId); 839 840 lStatus = chain->addEffect_l(effect); 841 if (lStatus != NO_ERROR) { 842 goto Exit; 843 } 844 effectCreated = true; 845 846 effect->setDevice(mOutDevice); 847 effect->setDevice(mInDevice); 848 effect->setMode(mAudioFlinger->getMode()); 849 effect->setAudioSource(mAudioSource); 850 } 851 // create effect handle and connect it to effect module 852 handle = new EffectHandle(effect, client, effectClient, priority); 853 lStatus = handle->initCheck(); 854 if (lStatus == OK) { 855 lStatus = effect->addHandle(handle.get()); 856 } 857 if (enabled != NULL) { 858 *enabled = (int)effect->isEnabled(); 859 } 860 } 861 862Exit: 863 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 864 Mutex::Autolock _l(mLock); 865 if (effectCreated) { 866 chain->removeEffect_l(effect); 867 } 868 if (effectRegistered) { 869 AudioSystem::unregisterEffect(effect->id()); 870 } 871 if (chainCreated) { 872 removeEffectChain_l(chain); 873 } 874 handle.clear(); 875 } 876 877 *status = lStatus; 878 return handle; 879} 880 881sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 882{ 883 Mutex::Autolock _l(mLock); 884 return getEffect_l(sessionId, effectId); 885} 886 887sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 888{ 889 sp<EffectChain> chain = getEffectChain_l(sessionId); 890 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 891} 892 893// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 894// PlaybackThread::mLock held 895status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 896{ 897 // check for existing effect chain with the requested audio session 898 int sessionId = effect->sessionId(); 899 sp<EffectChain> chain = getEffectChain_l(sessionId); 900 bool chainCreated = false; 901 902 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 903 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 904 this, effect->desc().name, effect->desc().flags); 905 906 if (chain == 0) { 907 // create a new chain for this session 908 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 909 chain = new EffectChain(this, sessionId); 910 addEffectChain_l(chain); 911 chain->setStrategy(getStrategyForSession_l(sessionId)); 912 chainCreated = true; 913 } 914 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 915 916 if (chain->getEffectFromId_l(effect->id()) != 0) { 917 ALOGW("addEffect_l() %p effect %s already present in chain %p", 918 this, effect->desc().name, chain.get()); 919 return BAD_VALUE; 920 } 921 922 effect->setOffloaded(mType == OFFLOAD, mId); 923 924 status_t status = chain->addEffect_l(effect); 925 if (status != NO_ERROR) { 926 if (chainCreated) { 927 removeEffectChain_l(chain); 928 } 929 return status; 930 } 931 932 effect->setDevice(mOutDevice); 933 effect->setDevice(mInDevice); 934 effect->setMode(mAudioFlinger->getMode()); 935 effect->setAudioSource(mAudioSource); 936 return NO_ERROR; 937} 938 939void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 940 941 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 942 effect_descriptor_t desc = effect->desc(); 943 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 944 detachAuxEffect_l(effect->id()); 945 } 946 947 sp<EffectChain> chain = effect->chain().promote(); 948 if (chain != 0) { 949 // remove effect chain if removing last effect 950 if (chain->removeEffect_l(effect) == 0) { 951 removeEffectChain_l(chain); 952 } 953 } else { 954 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 955 } 956} 957 958void AudioFlinger::ThreadBase::lockEffectChains_l( 959 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 960{ 961 effectChains = mEffectChains; 962 for (size_t i = 0; i < mEffectChains.size(); i++) { 963 mEffectChains[i]->lock(); 964 } 965} 966 967void AudioFlinger::ThreadBase::unlockEffectChains( 968 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 969{ 970 for (size_t i = 0; i < effectChains.size(); i++) { 971 effectChains[i]->unlock(); 972 } 973} 974 975sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 976{ 977 Mutex::Autolock _l(mLock); 978 return getEffectChain_l(sessionId); 979} 980 981sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 982{ 983 size_t size = mEffectChains.size(); 984 for (size_t i = 0; i < size; i++) { 985 if (mEffectChains[i]->sessionId() == sessionId) { 986 return mEffectChains[i]; 987 } 988 } 989 return 0; 990} 991 992void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 993{ 994 Mutex::Autolock _l(mLock); 995 size_t size = mEffectChains.size(); 996 for (size_t i = 0; i < size; i++) { 997 mEffectChains[i]->setMode_l(mode); 998 } 999} 1000 1001void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1002 EffectHandle *handle, 1003 bool unpinIfLast) { 1004 1005 Mutex::Autolock _l(mLock); 1006 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1007 // delete the effect module if removing last handle on it 1008 if (effect->removeHandle(handle) == 0) { 1009 if (!effect->isPinned() || unpinIfLast) { 1010 removeEffect_l(effect); 1011 AudioSystem::unregisterEffect(effect->id()); 1012 } 1013 } 1014} 1015 1016// ---------------------------------------------------------------------------- 1017// Playback 1018// ---------------------------------------------------------------------------- 1019 1020AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1021 AudioStreamOut* output, 1022 audio_io_handle_t id, 1023 audio_devices_t device, 1024 type_t type) 1025 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1026 mNormalFrameCount(0), mMixBuffer(NULL), 1027 mSuspended(0), mBytesWritten(0), 1028 mActiveTracksGeneration(0), 1029 // mStreamTypes[] initialized in constructor body 1030 mOutput(output), 1031 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1032 mMixerStatus(MIXER_IDLE), 1033 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1034 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1035 mBytesRemaining(0), 1036 mCurrentWriteLength(0), 1037 mUseAsyncWrite(false), 1038 mWriteAckSequence(0), 1039 mDrainSequence(0), 1040 mSignalPending(false), 1041 mScreenState(AudioFlinger::mScreenState), 1042 // index 0 is reserved for normal mixer's submix 1043 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1044 // mLatchD, mLatchQ, 1045 mLatchDValid(false), mLatchQValid(false) 1046{ 1047 snprintf(mName, kNameLength, "AudioOut_%X", id); 1048 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1049 1050 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1051 // it would be safer to explicitly pass initial masterVolume/masterMute as 1052 // parameter. 1053 // 1054 // If the HAL we are using has support for master volume or master mute, 1055 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1056 // and the mute set to false). 1057 mMasterVolume = audioFlinger->masterVolume_l(); 1058 mMasterMute = audioFlinger->masterMute_l(); 1059 if (mOutput && mOutput->audioHwDev) { 1060 if (mOutput->audioHwDev->canSetMasterVolume()) { 1061 mMasterVolume = 1.0; 1062 } 1063 1064 if (mOutput->audioHwDev->canSetMasterMute()) { 1065 mMasterMute = false; 1066 } 1067 } 1068 1069 readOutputParameters(); 1070 1071 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1072 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1073 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1074 stream = (audio_stream_type_t) (stream + 1)) { 1075 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1076 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1077 } 1078 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1079 // because mAudioFlinger doesn't have one to copy from 1080} 1081 1082AudioFlinger::PlaybackThread::~PlaybackThread() 1083{ 1084 mAudioFlinger->unregisterWriter(mNBLogWriter); 1085 delete[] mMixBuffer; 1086} 1087 1088void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1089{ 1090 dumpInternals(fd, args); 1091 dumpTracks(fd, args); 1092 dumpEffectChains(fd, args); 1093} 1094 1095void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1096{ 1097 const size_t SIZE = 256; 1098 char buffer[SIZE]; 1099 String8 result; 1100 1101 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1102 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1103 const stream_type_t *st = &mStreamTypes[i]; 1104 if (i > 0) { 1105 result.appendFormat(", "); 1106 } 1107 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1108 if (st->mute) { 1109 result.append("M"); 1110 } 1111 } 1112 result.append("\n"); 1113 write(fd, result.string(), result.length()); 1114 result.clear(); 1115 1116 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1117 result.append(buffer); 1118 Track::appendDumpHeader(result); 1119 for (size_t i = 0; i < mTracks.size(); ++i) { 1120 sp<Track> track = mTracks[i]; 1121 if (track != 0) { 1122 track->dump(buffer, SIZE); 1123 result.append(buffer); 1124 } 1125 } 1126 1127 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1128 result.append(buffer); 1129 Track::appendDumpHeader(result); 1130 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1131 sp<Track> track = mActiveTracks[i].promote(); 1132 if (track != 0) { 1133 track->dump(buffer, SIZE); 1134 result.append(buffer); 1135 } 1136 } 1137 write(fd, result.string(), result.size()); 1138 1139 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1140 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1141 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1142 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1143} 1144 1145void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1146{ 1147 const size_t SIZE = 256; 1148 char buffer[SIZE]; 1149 String8 result; 1150 1151 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1152 result.append(buffer); 1153 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1154 result.append(buffer); 1155 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1156 ns2ms(systemTime() - mLastWriteTime)); 1157 result.append(buffer); 1158 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1159 result.append(buffer); 1160 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1161 result.append(buffer); 1162 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1163 result.append(buffer); 1164 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1165 result.append(buffer); 1166 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1167 result.append(buffer); 1168 write(fd, result.string(), result.size()); 1169 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1170 1171 dumpBase(fd, args); 1172} 1173 1174// Thread virtuals 1175 1176void AudioFlinger::PlaybackThread::onFirstRef() 1177{ 1178 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1179} 1180 1181// ThreadBase virtuals 1182void AudioFlinger::PlaybackThread::preExit() 1183{ 1184 ALOGV(" preExit()"); 1185 // FIXME this is using hard-coded strings but in the future, this functionality will be 1186 // converted to use audio HAL extensions required to support tunneling 1187 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1188} 1189 1190// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1191sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1192 const sp<AudioFlinger::Client>& client, 1193 audio_stream_type_t streamType, 1194 uint32_t sampleRate, 1195 audio_format_t format, 1196 audio_channel_mask_t channelMask, 1197 size_t *pFrameCount, 1198 const sp<IMemory>& sharedBuffer, 1199 int sessionId, 1200 IAudioFlinger::track_flags_t *flags, 1201 pid_t tid, 1202 int uid, 1203 status_t *status) 1204{ 1205 size_t frameCount = *pFrameCount; 1206 sp<Track> track; 1207 status_t lStatus; 1208 1209 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1210 1211 // client expresses a preference for FAST, but we get the final say 1212 if (*flags & IAudioFlinger::TRACK_FAST) { 1213 if ( 1214 // not timed 1215 (!isTimed) && 1216 // either of these use cases: 1217 ( 1218 // use case 1: shared buffer with any frame count 1219 ( 1220 (sharedBuffer != 0) 1221 ) || 1222 // use case 2: callback handler and frame count is default or at least as large as HAL 1223 ( 1224 (tid != -1) && 1225 ((frameCount == 0) || 1226 (frameCount >= mFrameCount)) 1227 ) 1228 ) && 1229 // PCM data 1230 audio_is_linear_pcm(format) && 1231 // mono or stereo 1232 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1233 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1234#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1235 // hardware sample rate 1236 (sampleRate == mSampleRate) && 1237#endif 1238 // normal mixer has an associated fast mixer 1239 hasFastMixer() && 1240 // there are sufficient fast track slots available 1241 (mFastTrackAvailMask != 0) 1242 // FIXME test that MixerThread for this fast track has a capable output HAL 1243 // FIXME add a permission test also? 1244 ) { 1245 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1246 if (frameCount == 0) { 1247 frameCount = mFrameCount * kFastTrackMultiplier; 1248 } 1249 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1250 frameCount, mFrameCount); 1251 } else { 1252 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1253 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1254 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1255 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1256 audio_is_linear_pcm(format), 1257 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1258 *flags &= ~IAudioFlinger::TRACK_FAST; 1259 // For compatibility with AudioTrack calculation, buffer depth is forced 1260 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1261 // This is probably too conservative, but legacy application code may depend on it. 1262 // If you change this calculation, also review the start threshold which is related. 1263 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1264 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1265 if (minBufCount < 2) { 1266 minBufCount = 2; 1267 } 1268 size_t minFrameCount = mNormalFrameCount * minBufCount; 1269 if (frameCount < minFrameCount) { 1270 frameCount = minFrameCount; 1271 } 1272 } 1273 } 1274 *pFrameCount = frameCount; 1275 1276 if (mType == DIRECT) { 1277 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1278 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1279 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1280 "for output %p with format %d", 1281 sampleRate, format, channelMask, mOutput, mFormat); 1282 lStatus = BAD_VALUE; 1283 goto Exit; 1284 } 1285 } 1286 } else if (mType == OFFLOAD) { 1287 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1288 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1289 "for output %p with format %d", 1290 sampleRate, format, channelMask, mOutput, mFormat); 1291 lStatus = BAD_VALUE; 1292 goto Exit; 1293 } 1294 } else { 1295 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1296 ALOGE("createTrack_l() Bad parameter: format %d \"" 1297 "for output %p with format %d", 1298 format, mOutput, mFormat); 1299 lStatus = BAD_VALUE; 1300 goto Exit; 1301 } 1302 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1303 if (sampleRate > mSampleRate*2) { 1304 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1305 lStatus = BAD_VALUE; 1306 goto Exit; 1307 } 1308 } 1309 1310 lStatus = initCheck(); 1311 if (lStatus != NO_ERROR) { 1312 ALOGE("Audio driver not initialized."); 1313 goto Exit; 1314 } 1315 1316 { // scope for mLock 1317 Mutex::Autolock _l(mLock); 1318 1319 // all tracks in same audio session must share the same routing strategy otherwise 1320 // conflicts will happen when tracks are moved from one output to another by audio policy 1321 // manager 1322 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1323 for (size_t i = 0; i < mTracks.size(); ++i) { 1324 sp<Track> t = mTracks[i]; 1325 if (t != 0 && !t->isOutputTrack()) { 1326 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1327 if (sessionId == t->sessionId() && strategy != actual) { 1328 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1329 strategy, actual); 1330 lStatus = BAD_VALUE; 1331 goto Exit; 1332 } 1333 } 1334 } 1335 1336 if (!isTimed) { 1337 track = new Track(this, client, streamType, sampleRate, format, 1338 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1339 } else { 1340 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1341 channelMask, frameCount, sharedBuffer, sessionId, uid); 1342 } 1343 1344 // new Track always returns non-NULL, 1345 // but TimedTrack::create() is a factory that could fail by returning NULL 1346 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1347 if (lStatus != NO_ERROR) { 1348 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1349 // track must be cleared from the caller as the caller has the AF lock 1350 goto Exit; 1351 } 1352 1353 mTracks.add(track); 1354 1355 sp<EffectChain> chain = getEffectChain_l(sessionId); 1356 if (chain != 0) { 1357 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1358 track->setMainBuffer(chain->inBuffer()); 1359 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1360 chain->incTrackCnt(); 1361 } 1362 1363 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1364 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1365 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1366 // so ask activity manager to do this on our behalf 1367 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1368 } 1369 } 1370 1371 lStatus = NO_ERROR; 1372 1373Exit: 1374 *status = lStatus; 1375 return track; 1376} 1377 1378uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1379{ 1380 return latency; 1381} 1382 1383uint32_t AudioFlinger::PlaybackThread::latency() const 1384{ 1385 Mutex::Autolock _l(mLock); 1386 return latency_l(); 1387} 1388uint32_t AudioFlinger::PlaybackThread::latency_l() const 1389{ 1390 if (initCheck() == NO_ERROR) { 1391 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1392 } else { 1393 return 0; 1394 } 1395} 1396 1397void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1398{ 1399 Mutex::Autolock _l(mLock); 1400 // Don't apply master volume in SW if our HAL can do it for us. 1401 if (mOutput && mOutput->audioHwDev && 1402 mOutput->audioHwDev->canSetMasterVolume()) { 1403 mMasterVolume = 1.0; 1404 } else { 1405 mMasterVolume = value; 1406 } 1407} 1408 1409void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1410{ 1411 Mutex::Autolock _l(mLock); 1412 // Don't apply master mute in SW if our HAL can do it for us. 1413 if (mOutput && mOutput->audioHwDev && 1414 mOutput->audioHwDev->canSetMasterMute()) { 1415 mMasterMute = false; 1416 } else { 1417 mMasterMute = muted; 1418 } 1419} 1420 1421void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1422{ 1423 Mutex::Autolock _l(mLock); 1424 mStreamTypes[stream].volume = value; 1425 broadcast_l(); 1426} 1427 1428void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1429{ 1430 Mutex::Autolock _l(mLock); 1431 mStreamTypes[stream].mute = muted; 1432 broadcast_l(); 1433} 1434 1435float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1436{ 1437 Mutex::Autolock _l(mLock); 1438 return mStreamTypes[stream].volume; 1439} 1440 1441// addTrack_l() must be called with ThreadBase::mLock held 1442status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1443{ 1444 status_t status = ALREADY_EXISTS; 1445 1446 // set retry count for buffer fill 1447 track->mRetryCount = kMaxTrackStartupRetries; 1448 if (mActiveTracks.indexOf(track) < 0) { 1449 // the track is newly added, make sure it fills up all its 1450 // buffers before playing. This is to ensure the client will 1451 // effectively get the latency it requested. 1452 if (!track->isOutputTrack()) { 1453 TrackBase::track_state state = track->mState; 1454 mLock.unlock(); 1455 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1456 mLock.lock(); 1457 // abort track was stopped/paused while we released the lock 1458 if (state != track->mState) { 1459 if (status == NO_ERROR) { 1460 mLock.unlock(); 1461 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1462 mLock.lock(); 1463 } 1464 return INVALID_OPERATION; 1465 } 1466 // abort if start is rejected by audio policy manager 1467 if (status != NO_ERROR) { 1468 return PERMISSION_DENIED; 1469 } 1470#ifdef ADD_BATTERY_DATA 1471 // to track the speaker usage 1472 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1473#endif 1474 } 1475 1476 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1477 track->mResetDone = false; 1478 track->mPresentationCompleteFrames = 0; 1479 mActiveTracks.add(track); 1480 mWakeLockUids.add(track->uid()); 1481 mActiveTracksGeneration++; 1482 mLatestActiveTrack = track; 1483 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1484 if (chain != 0) { 1485 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1486 track->sessionId()); 1487 chain->incActiveTrackCnt(); 1488 } 1489 1490 status = NO_ERROR; 1491 } 1492 1493 ALOGV("signal playback thread"); 1494 broadcast_l(); 1495 1496 return status; 1497} 1498 1499bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1500{ 1501 track->terminate(); 1502 // active tracks are removed by threadLoop() 1503 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1504 track->mState = TrackBase::STOPPED; 1505 if (!trackActive) { 1506 removeTrack_l(track); 1507 } else if (track->isFastTrack() || track->isOffloaded()) { 1508 track->mState = TrackBase::STOPPING_1; 1509 } 1510 1511 return trackActive; 1512} 1513 1514void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1515{ 1516 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1517 mTracks.remove(track); 1518 deleteTrackName_l(track->name()); 1519 // redundant as track is about to be destroyed, for dumpsys only 1520 track->mName = -1; 1521 if (track->isFastTrack()) { 1522 int index = track->mFastIndex; 1523 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1524 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1525 mFastTrackAvailMask |= 1 << index; 1526 // redundant as track is about to be destroyed, for dumpsys only 1527 track->mFastIndex = -1; 1528 } 1529 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1530 if (chain != 0) { 1531 chain->decTrackCnt(); 1532 } 1533} 1534 1535void AudioFlinger::PlaybackThread::broadcast_l() 1536{ 1537 // Thread could be blocked waiting for async 1538 // so signal it to handle state changes immediately 1539 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1540 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1541 mSignalPending = true; 1542 mWaitWorkCV.broadcast(); 1543} 1544 1545String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1546{ 1547 Mutex::Autolock _l(mLock); 1548 if (initCheck() != NO_ERROR) { 1549 return String8(); 1550 } 1551 1552 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1553 const String8 out_s8(s); 1554 free(s); 1555 return out_s8; 1556} 1557 1558// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1559void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1560 AudioSystem::OutputDescriptor desc; 1561 void *param2 = NULL; 1562 1563 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1564 param); 1565 1566 switch (event) { 1567 case AudioSystem::OUTPUT_OPENED: 1568 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1569 desc.channelMask = mChannelMask; 1570 desc.samplingRate = mSampleRate; 1571 desc.format = mFormat; 1572 desc.frameCount = mNormalFrameCount; // FIXME see 1573 // AudioFlinger::frameCount(audio_io_handle_t) 1574 desc.latency = latency(); 1575 param2 = &desc; 1576 break; 1577 1578 case AudioSystem::STREAM_CONFIG_CHANGED: 1579 param2 = ¶m; 1580 case AudioSystem::OUTPUT_CLOSED: 1581 default: 1582 break; 1583 } 1584 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1585} 1586 1587void AudioFlinger::PlaybackThread::writeCallback() 1588{ 1589 ALOG_ASSERT(mCallbackThread != 0); 1590 mCallbackThread->resetWriteBlocked(); 1591} 1592 1593void AudioFlinger::PlaybackThread::drainCallback() 1594{ 1595 ALOG_ASSERT(mCallbackThread != 0); 1596 mCallbackThread->resetDraining(); 1597} 1598 1599void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1600{ 1601 Mutex::Autolock _l(mLock); 1602 // reject out of sequence requests 1603 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1604 mWriteAckSequence &= ~1; 1605 mWaitWorkCV.signal(); 1606 } 1607} 1608 1609void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1610{ 1611 Mutex::Autolock _l(mLock); 1612 // reject out of sequence requests 1613 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1614 mDrainSequence &= ~1; 1615 mWaitWorkCV.signal(); 1616 } 1617} 1618 1619// static 1620int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1621 void *param __unused, 1622 void *cookie) 1623{ 1624 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1625 ALOGV("asyncCallback() event %d", event); 1626 switch (event) { 1627 case STREAM_CBK_EVENT_WRITE_READY: 1628 me->writeCallback(); 1629 break; 1630 case STREAM_CBK_EVENT_DRAIN_READY: 1631 me->drainCallback(); 1632 break; 1633 default: 1634 ALOGW("asyncCallback() unknown event %d", event); 1635 break; 1636 } 1637 return 0; 1638} 1639 1640void AudioFlinger::PlaybackThread::readOutputParameters() 1641{ 1642 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1643 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1644 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1645 if (!audio_is_output_channel(mChannelMask)) { 1646 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1647 } 1648 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1649 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1650 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1651 } 1652 mChannelCount = popcount(mChannelMask); 1653 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1654 if (!audio_is_valid_format(mFormat)) { 1655 LOG_FATAL("HAL format %d not valid for output", mFormat); 1656 } 1657 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1658 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1659 mFormat); 1660 } 1661 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1662 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1663 mFrameCount = mBufferSize / mFrameSize; 1664 if (mFrameCount & 15) { 1665 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1666 mFrameCount); 1667 } 1668 1669 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1670 (mOutput->stream->set_callback != NULL)) { 1671 if (mOutput->stream->set_callback(mOutput->stream, 1672 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1673 mUseAsyncWrite = true; 1674 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1675 } 1676 } 1677 1678 // Calculate size of normal mix buffer relative to the HAL output buffer size 1679 double multiplier = 1.0; 1680 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1681 kUseFastMixer == FastMixer_Dynamic)) { 1682 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1683 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1684 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1685 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1686 maxNormalFrameCount = maxNormalFrameCount & ~15; 1687 if (maxNormalFrameCount < minNormalFrameCount) { 1688 maxNormalFrameCount = minNormalFrameCount; 1689 } 1690 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1691 if (multiplier <= 1.0) { 1692 multiplier = 1.0; 1693 } else if (multiplier <= 2.0) { 1694 if (2 * mFrameCount <= maxNormalFrameCount) { 1695 multiplier = 2.0; 1696 } else { 1697 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1698 } 1699 } else { 1700 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1701 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1702 // track, but we sometimes have to do this to satisfy the maximum frame count 1703 // constraint) 1704 // FIXME this rounding up should not be done if no HAL SRC 1705 uint32_t truncMult = (uint32_t) multiplier; 1706 if ((truncMult & 1)) { 1707 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1708 ++truncMult; 1709 } 1710 } 1711 multiplier = (double) truncMult; 1712 } 1713 } 1714 mNormalFrameCount = multiplier * mFrameCount; 1715 // round up to nearest 16 frames to satisfy AudioMixer 1716 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1717 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1718 mNormalFrameCount); 1719 1720 delete[] mMixBuffer; 1721 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1722 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1723 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1724 memset(mMixBuffer, 0, normalBufferSize); 1725 1726 // force reconfiguration of effect chains and engines to take new buffer size and audio 1727 // parameters into account 1728 // Note that mLock is not held when readOutputParameters() is called from the constructor 1729 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1730 // matter. 1731 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1732 Vector< sp<EffectChain> > effectChains = mEffectChains; 1733 for (size_t i = 0; i < effectChains.size(); i ++) { 1734 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1735 } 1736} 1737 1738 1739status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1740{ 1741 if (halFrames == NULL || dspFrames == NULL) { 1742 return BAD_VALUE; 1743 } 1744 Mutex::Autolock _l(mLock); 1745 if (initCheck() != NO_ERROR) { 1746 return INVALID_OPERATION; 1747 } 1748 size_t framesWritten = mBytesWritten / mFrameSize; 1749 *halFrames = framesWritten; 1750 1751 if (isSuspended()) { 1752 // return an estimation of rendered frames when the output is suspended 1753 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1754 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1755 return NO_ERROR; 1756 } else { 1757 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1758 } 1759} 1760 1761uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1762{ 1763 Mutex::Autolock _l(mLock); 1764 uint32_t result = 0; 1765 if (getEffectChain_l(sessionId) != 0) { 1766 result = EFFECT_SESSION; 1767 } 1768 1769 for (size_t i = 0; i < mTracks.size(); ++i) { 1770 sp<Track> track = mTracks[i]; 1771 if (sessionId == track->sessionId() && !track->isInvalid()) { 1772 result |= TRACK_SESSION; 1773 break; 1774 } 1775 } 1776 1777 return result; 1778} 1779 1780uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1781{ 1782 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1783 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1784 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1785 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1786 } 1787 for (size_t i = 0; i < mTracks.size(); i++) { 1788 sp<Track> track = mTracks[i]; 1789 if (sessionId == track->sessionId() && !track->isInvalid()) { 1790 return AudioSystem::getStrategyForStream(track->streamType()); 1791 } 1792 } 1793 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1794} 1795 1796 1797AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1798{ 1799 Mutex::Autolock _l(mLock); 1800 return mOutput; 1801} 1802 1803AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1804{ 1805 Mutex::Autolock _l(mLock); 1806 AudioStreamOut *output = mOutput; 1807 mOutput = NULL; 1808 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1809 // must push a NULL and wait for ack 1810 mOutputSink.clear(); 1811 mPipeSink.clear(); 1812 mNormalSink.clear(); 1813 return output; 1814} 1815 1816// this method must always be called either with ThreadBase mLock held or inside the thread loop 1817audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1818{ 1819 if (mOutput == NULL) { 1820 return NULL; 1821 } 1822 return &mOutput->stream->common; 1823} 1824 1825uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1826{ 1827 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1828} 1829 1830status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1831{ 1832 if (!isValidSyncEvent(event)) { 1833 return BAD_VALUE; 1834 } 1835 1836 Mutex::Autolock _l(mLock); 1837 1838 for (size_t i = 0; i < mTracks.size(); ++i) { 1839 sp<Track> track = mTracks[i]; 1840 if (event->triggerSession() == track->sessionId()) { 1841 (void) track->setSyncEvent(event); 1842 return NO_ERROR; 1843 } 1844 } 1845 1846 return NAME_NOT_FOUND; 1847} 1848 1849bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1850{ 1851 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1852} 1853 1854void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1855 const Vector< sp<Track> >& tracksToRemove) 1856{ 1857 size_t count = tracksToRemove.size(); 1858 if (count > 0) { 1859 for (size_t i = 0 ; i < count ; i++) { 1860 const sp<Track>& track = tracksToRemove.itemAt(i); 1861 if (!track->isOutputTrack()) { 1862 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1863#ifdef ADD_BATTERY_DATA 1864 // to track the speaker usage 1865 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1866#endif 1867 if (track->isTerminated()) { 1868 AudioSystem::releaseOutput(mId); 1869 } 1870 } 1871 } 1872 } 1873} 1874 1875void AudioFlinger::PlaybackThread::checkSilentMode_l() 1876{ 1877 if (!mMasterMute) { 1878 char value[PROPERTY_VALUE_MAX]; 1879 if (property_get("ro.audio.silent", value, "0") > 0) { 1880 char *endptr; 1881 unsigned long ul = strtoul(value, &endptr, 0); 1882 if (*endptr == '\0' && ul != 0) { 1883 ALOGD("Silence is golden"); 1884 // The setprop command will not allow a property to be changed after 1885 // the first time it is set, so we don't have to worry about un-muting. 1886 setMasterMute_l(true); 1887 } 1888 } 1889 } 1890} 1891 1892// shared by MIXER and DIRECT, overridden by DUPLICATING 1893ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1894{ 1895 // FIXME rewrite to reduce number of system calls 1896 mLastWriteTime = systemTime(); 1897 mInWrite = true; 1898 ssize_t bytesWritten; 1899 1900 // If an NBAIO sink is present, use it to write the normal mixer's submix 1901 if (mNormalSink != 0) { 1902#define mBitShift 2 // FIXME 1903 size_t count = mBytesRemaining >> mBitShift; 1904 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1905 ATRACE_BEGIN("write"); 1906 // update the setpoint when AudioFlinger::mScreenState changes 1907 uint32_t screenState = AudioFlinger::mScreenState; 1908 if (screenState != mScreenState) { 1909 mScreenState = screenState; 1910 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1911 if (pipe != NULL) { 1912 pipe->setAvgFrames((mScreenState & 1) ? 1913 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1914 } 1915 } 1916 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1917 ATRACE_END(); 1918 if (framesWritten > 0) { 1919 bytesWritten = framesWritten << mBitShift; 1920 } else { 1921 bytesWritten = framesWritten; 1922 } 1923 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1924 if (status == NO_ERROR) { 1925 size_t totalFramesWritten = mNormalSink->framesWritten(); 1926 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1927 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1928 mLatchDValid = true; 1929 } 1930 } 1931 // otherwise use the HAL / AudioStreamOut directly 1932 } else { 1933 // Direct output and offload threads 1934 size_t offset = (mCurrentWriteLength - mBytesRemaining); 1935 if (mUseAsyncWrite) { 1936 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1937 mWriteAckSequence += 2; 1938 mWriteAckSequence |= 1; 1939 ALOG_ASSERT(mCallbackThread != 0); 1940 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1941 } 1942 // FIXME We should have an implementation of timestamps for direct output threads. 1943 // They are used e.g for multichannel PCM playback over HDMI. 1944 bytesWritten = mOutput->stream->write(mOutput->stream, 1945 (char *)mMixBuffer + offset, mBytesRemaining); 1946 if (mUseAsyncWrite && 1947 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1948 // do not wait for async callback in case of error of full write 1949 mWriteAckSequence &= ~1; 1950 ALOG_ASSERT(mCallbackThread != 0); 1951 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1952 } 1953 } 1954 1955 mNumWrites++; 1956 mInWrite = false; 1957 mStandby = false; 1958 return bytesWritten; 1959} 1960 1961void AudioFlinger::PlaybackThread::threadLoop_drain() 1962{ 1963 if (mOutput->stream->drain) { 1964 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1965 if (mUseAsyncWrite) { 1966 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1967 mDrainSequence |= 1; 1968 ALOG_ASSERT(mCallbackThread != 0); 1969 mCallbackThread->setDraining(mDrainSequence); 1970 } 1971 mOutput->stream->drain(mOutput->stream, 1972 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1973 : AUDIO_DRAIN_ALL); 1974 } 1975} 1976 1977void AudioFlinger::PlaybackThread::threadLoop_exit() 1978{ 1979 // Default implementation has nothing to do 1980} 1981 1982/* 1983The derived values that are cached: 1984 - mixBufferSize from frame count * frame size 1985 - activeSleepTime from activeSleepTimeUs() 1986 - idleSleepTime from idleSleepTimeUs() 1987 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1988 - maxPeriod from frame count and sample rate (MIXER only) 1989 1990The parameters that affect these derived values are: 1991 - frame count 1992 - frame size 1993 - sample rate 1994 - device type: A2DP or not 1995 - device latency 1996 - format: PCM or not 1997 - active sleep time 1998 - idle sleep time 1999*/ 2000 2001void AudioFlinger::PlaybackThread::cacheParameters_l() 2002{ 2003 mixBufferSize = mNormalFrameCount * mFrameSize; 2004 activeSleepTime = activeSleepTimeUs(); 2005 idleSleepTime = idleSleepTimeUs(); 2006} 2007 2008void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2009{ 2010 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2011 this, streamType, mTracks.size()); 2012 Mutex::Autolock _l(mLock); 2013 2014 size_t size = mTracks.size(); 2015 for (size_t i = 0; i < size; i++) { 2016 sp<Track> t = mTracks[i]; 2017 if (t->streamType() == streamType) { 2018 t->invalidate(); 2019 } 2020 } 2021} 2022 2023status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2024{ 2025 int session = chain->sessionId(); 2026 int16_t *buffer = mMixBuffer; 2027 bool ownsBuffer = false; 2028 2029 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2030 if (session > 0) { 2031 // Only one effect chain can be present in direct output thread and it uses 2032 // the mix buffer as input 2033 if (mType != DIRECT) { 2034 size_t numSamples = mNormalFrameCount * mChannelCount; 2035 buffer = new int16_t[numSamples]; 2036 memset(buffer, 0, numSamples * sizeof(int16_t)); 2037 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2038 ownsBuffer = true; 2039 } 2040 2041 // Attach all tracks with same session ID to this chain. 2042 for (size_t i = 0; i < mTracks.size(); ++i) { 2043 sp<Track> track = mTracks[i]; 2044 if (session == track->sessionId()) { 2045 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2046 buffer); 2047 track->setMainBuffer(buffer); 2048 chain->incTrackCnt(); 2049 } 2050 } 2051 2052 // indicate all active tracks in the chain 2053 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2054 sp<Track> track = mActiveTracks[i].promote(); 2055 if (track == 0) { 2056 continue; 2057 } 2058 if (session == track->sessionId()) { 2059 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2060 chain->incActiveTrackCnt(); 2061 } 2062 } 2063 } 2064 2065 chain->setInBuffer(buffer, ownsBuffer); 2066 chain->setOutBuffer(mMixBuffer); 2067 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2068 // chains list in order to be processed last as it contains output stage effects 2069 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2070 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2071 // after track specific effects and before output stage 2072 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2073 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2074 // Effect chain for other sessions are inserted at beginning of effect 2075 // chains list to be processed before output mix effects. Relative order between other 2076 // sessions is not important 2077 size_t size = mEffectChains.size(); 2078 size_t i = 0; 2079 for (i = 0; i < size; i++) { 2080 if (mEffectChains[i]->sessionId() < session) { 2081 break; 2082 } 2083 } 2084 mEffectChains.insertAt(chain, i); 2085 checkSuspendOnAddEffectChain_l(chain); 2086 2087 return NO_ERROR; 2088} 2089 2090size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2091{ 2092 int session = chain->sessionId(); 2093 2094 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2095 2096 for (size_t i = 0; i < mEffectChains.size(); i++) { 2097 if (chain == mEffectChains[i]) { 2098 mEffectChains.removeAt(i); 2099 // detach all active tracks from the chain 2100 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2101 sp<Track> track = mActiveTracks[i].promote(); 2102 if (track == 0) { 2103 continue; 2104 } 2105 if (session == track->sessionId()) { 2106 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2107 chain.get(), session); 2108 chain->decActiveTrackCnt(); 2109 } 2110 } 2111 2112 // detach all tracks with same session ID from this chain 2113 for (size_t i = 0; i < mTracks.size(); ++i) { 2114 sp<Track> track = mTracks[i]; 2115 if (session == track->sessionId()) { 2116 track->setMainBuffer(mMixBuffer); 2117 chain->decTrackCnt(); 2118 } 2119 } 2120 break; 2121 } 2122 } 2123 return mEffectChains.size(); 2124} 2125 2126status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2127 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2128{ 2129 Mutex::Autolock _l(mLock); 2130 return attachAuxEffect_l(track, EffectId); 2131} 2132 2133status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2134 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2135{ 2136 status_t status = NO_ERROR; 2137 2138 if (EffectId == 0) { 2139 track->setAuxBuffer(0, NULL); 2140 } else { 2141 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2142 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2143 if (effect != 0) { 2144 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2145 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2146 } else { 2147 status = INVALID_OPERATION; 2148 } 2149 } else { 2150 status = BAD_VALUE; 2151 } 2152 } 2153 return status; 2154} 2155 2156void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2157{ 2158 for (size_t i = 0; i < mTracks.size(); ++i) { 2159 sp<Track> track = mTracks[i]; 2160 if (track->auxEffectId() == effectId) { 2161 attachAuxEffect_l(track, 0); 2162 } 2163 } 2164} 2165 2166bool AudioFlinger::PlaybackThread::threadLoop() 2167{ 2168 Vector< sp<Track> > tracksToRemove; 2169 2170 standbyTime = systemTime(); 2171 2172 // MIXER 2173 nsecs_t lastWarning = 0; 2174 2175 // DUPLICATING 2176 // FIXME could this be made local to while loop? 2177 writeFrames = 0; 2178 2179 int lastGeneration = 0; 2180 2181 cacheParameters_l(); 2182 sleepTime = idleSleepTime; 2183 2184 if (mType == MIXER) { 2185 sleepTimeShift = 0; 2186 } 2187 2188 CpuStats cpuStats; 2189 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2190 2191 acquireWakeLock(); 2192 2193 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2194 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2195 // and then that string will be logged at the next convenient opportunity. 2196 const char *logString = NULL; 2197 2198 checkSilentMode_l(); 2199 2200 while (!exitPending()) 2201 { 2202 cpuStats.sample(myName); 2203 2204 Vector< sp<EffectChain> > effectChains; 2205 2206 processConfigEvents(); 2207 2208 { // scope for mLock 2209 2210 Mutex::Autolock _l(mLock); 2211 2212 if (logString != NULL) { 2213 mNBLogWriter->logTimestamp(); 2214 mNBLogWriter->log(logString); 2215 logString = NULL; 2216 } 2217 2218 if (mLatchDValid) { 2219 mLatchQ = mLatchD; 2220 mLatchDValid = false; 2221 mLatchQValid = true; 2222 } 2223 2224 if (checkForNewParameters_l()) { 2225 cacheParameters_l(); 2226 } 2227 2228 saveOutputTracks(); 2229 if (mSignalPending) { 2230 // A signal was raised while we were unlocked 2231 mSignalPending = false; 2232 } else if (waitingAsyncCallback_l()) { 2233 if (exitPending()) { 2234 break; 2235 } 2236 releaseWakeLock_l(); 2237 mWakeLockUids.clear(); 2238 mActiveTracksGeneration++; 2239 ALOGV("wait async completion"); 2240 mWaitWorkCV.wait(mLock); 2241 ALOGV("async completion/wake"); 2242 acquireWakeLock_l(); 2243 standbyTime = systemTime() + standbyDelay; 2244 sleepTime = 0; 2245 2246 continue; 2247 } 2248 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2249 isSuspended()) { 2250 // put audio hardware into standby after short delay 2251 if (shouldStandby_l()) { 2252 2253 threadLoop_standby(); 2254 2255 mStandby = true; 2256 } 2257 2258 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2259 // we're about to wait, flush the binder command buffer 2260 IPCThreadState::self()->flushCommands(); 2261 2262 clearOutputTracks(); 2263 2264 if (exitPending()) { 2265 break; 2266 } 2267 2268 releaseWakeLock_l(); 2269 mWakeLockUids.clear(); 2270 mActiveTracksGeneration++; 2271 // wait until we have something to do... 2272 ALOGV("%s going to sleep", myName.string()); 2273 mWaitWorkCV.wait(mLock); 2274 ALOGV("%s waking up", myName.string()); 2275 acquireWakeLock_l(); 2276 2277 mMixerStatus = MIXER_IDLE; 2278 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2279 mBytesWritten = 0; 2280 mBytesRemaining = 0; 2281 checkSilentMode_l(); 2282 2283 standbyTime = systemTime() + standbyDelay; 2284 sleepTime = idleSleepTime; 2285 if (mType == MIXER) { 2286 sleepTimeShift = 0; 2287 } 2288 2289 continue; 2290 } 2291 } 2292 // mMixerStatusIgnoringFastTracks is also updated internally 2293 mMixerStatus = prepareTracks_l(&tracksToRemove); 2294 2295 // compare with previously applied list 2296 if (lastGeneration != mActiveTracksGeneration) { 2297 // update wakelock 2298 updateWakeLockUids_l(mWakeLockUids); 2299 lastGeneration = mActiveTracksGeneration; 2300 } 2301 2302 // prevent any changes in effect chain list and in each effect chain 2303 // during mixing and effect process as the audio buffers could be deleted 2304 // or modified if an effect is created or deleted 2305 lockEffectChains_l(effectChains); 2306 } // mLock scope ends 2307 2308 if (mBytesRemaining == 0) { 2309 mCurrentWriteLength = 0; 2310 if (mMixerStatus == MIXER_TRACKS_READY) { 2311 // threadLoop_mix() sets mCurrentWriteLength 2312 threadLoop_mix(); 2313 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2314 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2315 // threadLoop_sleepTime sets sleepTime to 0 if data 2316 // must be written to HAL 2317 threadLoop_sleepTime(); 2318 if (sleepTime == 0) { 2319 mCurrentWriteLength = mixBufferSize; 2320 } 2321 } 2322 mBytesRemaining = mCurrentWriteLength; 2323 if (isSuspended()) { 2324 sleepTime = suspendSleepTimeUs(); 2325 // simulate write to HAL when suspended 2326 mBytesWritten += mixBufferSize; 2327 mBytesRemaining = 0; 2328 } 2329 2330 // only process effects if we're going to write 2331 if (sleepTime == 0 && mType != OFFLOAD) { 2332 for (size_t i = 0; i < effectChains.size(); i ++) { 2333 effectChains[i]->process_l(); 2334 } 2335 } 2336 } 2337 // Process effect chains for offloaded thread even if no audio 2338 // was read from audio track: process only updates effect state 2339 // and thus does have to be synchronized with audio writes but may have 2340 // to be called while waiting for async write callback 2341 if (mType == OFFLOAD) { 2342 for (size_t i = 0; i < effectChains.size(); i ++) { 2343 effectChains[i]->process_l(); 2344 } 2345 } 2346 2347 // enable changes in effect chain 2348 unlockEffectChains(effectChains); 2349 2350 if (!waitingAsyncCallback()) { 2351 // sleepTime == 0 means we must write to audio hardware 2352 if (sleepTime == 0) { 2353 if (mBytesRemaining) { 2354 ssize_t ret = threadLoop_write(); 2355 if (ret < 0) { 2356 mBytesRemaining = 0; 2357 } else { 2358 mBytesWritten += ret; 2359 mBytesRemaining -= ret; 2360 } 2361 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2362 (mMixerStatus == MIXER_DRAIN_ALL)) { 2363 threadLoop_drain(); 2364 } 2365if (mType == MIXER) { 2366 // write blocked detection 2367 nsecs_t now = systemTime(); 2368 nsecs_t delta = now - mLastWriteTime; 2369 if (!mStandby && delta > maxPeriod) { 2370 mNumDelayedWrites++; 2371 if ((now - lastWarning) > kWarningThrottleNs) { 2372 ATRACE_NAME("underrun"); 2373 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2374 ns2ms(delta), mNumDelayedWrites, this); 2375 lastWarning = now; 2376 } 2377 } 2378} 2379 2380 } else { 2381 usleep(sleepTime); 2382 } 2383 } 2384 2385 // Finally let go of removed track(s), without the lock held 2386 // since we can't guarantee the destructors won't acquire that 2387 // same lock. This will also mutate and push a new fast mixer state. 2388 threadLoop_removeTracks(tracksToRemove); 2389 tracksToRemove.clear(); 2390 2391 // FIXME I don't understand the need for this here; 2392 // it was in the original code but maybe the 2393 // assignment in saveOutputTracks() makes this unnecessary? 2394 clearOutputTracks(); 2395 2396 // Effect chains will be actually deleted here if they were removed from 2397 // mEffectChains list during mixing or effects processing 2398 effectChains.clear(); 2399 2400 // FIXME Note that the above .clear() is no longer necessary since effectChains 2401 // is now local to this block, but will keep it for now (at least until merge done). 2402 } 2403 2404 threadLoop_exit(); 2405 2406 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2407 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2408 // put output stream into standby mode 2409 if (!mStandby) { 2410 mOutput->stream->common.standby(&mOutput->stream->common); 2411 } 2412 } 2413 2414 releaseWakeLock(); 2415 mWakeLockUids.clear(); 2416 mActiveTracksGeneration++; 2417 2418 ALOGV("Thread %p type %d exiting", this, mType); 2419 return false; 2420} 2421 2422// removeTracks_l() must be called with ThreadBase::mLock held 2423void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2424{ 2425 size_t count = tracksToRemove.size(); 2426 if (count > 0) { 2427 for (size_t i=0 ; i<count ; i++) { 2428 const sp<Track>& track = tracksToRemove.itemAt(i); 2429 mActiveTracks.remove(track); 2430 mWakeLockUids.remove(track->uid()); 2431 mActiveTracksGeneration++; 2432 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2433 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2434 if (chain != 0) { 2435 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2436 track->sessionId()); 2437 chain->decActiveTrackCnt(); 2438 } 2439 if (track->isTerminated()) { 2440 removeTrack_l(track); 2441 } 2442 } 2443 } 2444 2445} 2446 2447status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2448{ 2449 if (mNormalSink != 0) { 2450 return mNormalSink->getTimestamp(timestamp); 2451 } 2452 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2453 uint64_t position64; 2454 int ret = mOutput->stream->get_presentation_position( 2455 mOutput->stream, &position64, ×tamp.mTime); 2456 if (ret == 0) { 2457 timestamp.mPosition = (uint32_t)position64; 2458 return NO_ERROR; 2459 } 2460 } 2461 return INVALID_OPERATION; 2462} 2463// ---------------------------------------------------------------------------- 2464 2465AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2466 audio_io_handle_t id, audio_devices_t device, type_t type) 2467 : PlaybackThread(audioFlinger, output, id, device, type), 2468 // mAudioMixer below 2469 // mFastMixer below 2470 mFastMixerFutex(0) 2471 // mOutputSink below 2472 // mPipeSink below 2473 // mNormalSink below 2474{ 2475 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2476 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2477 "mFrameCount=%d, mNormalFrameCount=%d", 2478 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2479 mNormalFrameCount); 2480 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2481 2482 // FIXME - Current mixer implementation only supports stereo output 2483 if (mChannelCount != FCC_2) { 2484 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2485 } 2486 2487 // create an NBAIO sink for the HAL output stream, and negotiate 2488 mOutputSink = new AudioStreamOutSink(output->stream); 2489 size_t numCounterOffers = 0; 2490 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2491 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2492 ALOG_ASSERT(index == 0); 2493 2494 // initialize fast mixer depending on configuration 2495 bool initFastMixer; 2496 switch (kUseFastMixer) { 2497 case FastMixer_Never: 2498 initFastMixer = false; 2499 break; 2500 case FastMixer_Always: 2501 initFastMixer = true; 2502 break; 2503 case FastMixer_Static: 2504 case FastMixer_Dynamic: 2505 initFastMixer = mFrameCount < mNormalFrameCount; 2506 break; 2507 } 2508 if (initFastMixer) { 2509 2510 // create a MonoPipe to connect our submix to FastMixer 2511 NBAIO_Format format = mOutputSink->format(); 2512 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2513 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2514 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2515 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2516 const NBAIO_Format offers[1] = {format}; 2517 size_t numCounterOffers = 0; 2518 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2519 ALOG_ASSERT(index == 0); 2520 monoPipe->setAvgFrames((mScreenState & 1) ? 2521 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2522 mPipeSink = monoPipe; 2523 2524#ifdef TEE_SINK 2525 if (mTeeSinkOutputEnabled) { 2526 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2527 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2528 numCounterOffers = 0; 2529 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2530 ALOG_ASSERT(index == 0); 2531 mTeeSink = teeSink; 2532 PipeReader *teeSource = new PipeReader(*teeSink); 2533 numCounterOffers = 0; 2534 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2535 ALOG_ASSERT(index == 0); 2536 mTeeSource = teeSource; 2537 } 2538#endif 2539 2540 // create fast mixer and configure it initially with just one fast track for our submix 2541 mFastMixer = new FastMixer(); 2542 FastMixerStateQueue *sq = mFastMixer->sq(); 2543#ifdef STATE_QUEUE_DUMP 2544 sq->setObserverDump(&mStateQueueObserverDump); 2545 sq->setMutatorDump(&mStateQueueMutatorDump); 2546#endif 2547 FastMixerState *state = sq->begin(); 2548 FastTrack *fastTrack = &state->mFastTracks[0]; 2549 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2550 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2551 fastTrack->mVolumeProvider = NULL; 2552 fastTrack->mGeneration++; 2553 state->mFastTracksGen++; 2554 state->mTrackMask = 1; 2555 // fast mixer will use the HAL output sink 2556 state->mOutputSink = mOutputSink.get(); 2557 state->mOutputSinkGen++; 2558 state->mFrameCount = mFrameCount; 2559 state->mCommand = FastMixerState::COLD_IDLE; 2560 // already done in constructor initialization list 2561 //mFastMixerFutex = 0; 2562 state->mColdFutexAddr = &mFastMixerFutex; 2563 state->mColdGen++; 2564 state->mDumpState = &mFastMixerDumpState; 2565#ifdef TEE_SINK 2566 state->mTeeSink = mTeeSink.get(); 2567#endif 2568 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2569 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2570 sq->end(); 2571 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2572 2573 // start the fast mixer 2574 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2575 pid_t tid = mFastMixer->getTid(); 2576 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2577 if (err != 0) { 2578 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2579 kPriorityFastMixer, getpid_cached, tid, err); 2580 } 2581 2582#ifdef AUDIO_WATCHDOG 2583 // create and start the watchdog 2584 mAudioWatchdog = new AudioWatchdog(); 2585 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2586 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2587 tid = mAudioWatchdog->getTid(); 2588 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2589 if (err != 0) { 2590 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2591 kPriorityFastMixer, getpid_cached, tid, err); 2592 } 2593#endif 2594 2595 } else { 2596 mFastMixer = NULL; 2597 } 2598 2599 switch (kUseFastMixer) { 2600 case FastMixer_Never: 2601 case FastMixer_Dynamic: 2602 mNormalSink = mOutputSink; 2603 break; 2604 case FastMixer_Always: 2605 mNormalSink = mPipeSink; 2606 break; 2607 case FastMixer_Static: 2608 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2609 break; 2610 } 2611} 2612 2613AudioFlinger::MixerThread::~MixerThread() 2614{ 2615 if (mFastMixer != NULL) { 2616 FastMixerStateQueue *sq = mFastMixer->sq(); 2617 FastMixerState *state = sq->begin(); 2618 if (state->mCommand == FastMixerState::COLD_IDLE) { 2619 int32_t old = android_atomic_inc(&mFastMixerFutex); 2620 if (old == -1) { 2621 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2622 } 2623 } 2624 state->mCommand = FastMixerState::EXIT; 2625 sq->end(); 2626 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2627 mFastMixer->join(); 2628 // Though the fast mixer thread has exited, it's state queue is still valid. 2629 // We'll use that extract the final state which contains one remaining fast track 2630 // corresponding to our sub-mix. 2631 state = sq->begin(); 2632 ALOG_ASSERT(state->mTrackMask == 1); 2633 FastTrack *fastTrack = &state->mFastTracks[0]; 2634 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2635 delete fastTrack->mBufferProvider; 2636 sq->end(false /*didModify*/); 2637 delete mFastMixer; 2638#ifdef AUDIO_WATCHDOG 2639 if (mAudioWatchdog != 0) { 2640 mAudioWatchdog->requestExit(); 2641 mAudioWatchdog->requestExitAndWait(); 2642 mAudioWatchdog.clear(); 2643 } 2644#endif 2645 } 2646 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2647 delete mAudioMixer; 2648} 2649 2650 2651uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2652{ 2653 if (mFastMixer != NULL) { 2654 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2655 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2656 } 2657 return latency; 2658} 2659 2660 2661void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2662{ 2663 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2664} 2665 2666ssize_t AudioFlinger::MixerThread::threadLoop_write() 2667{ 2668 // FIXME we should only do one push per cycle; confirm this is true 2669 // Start the fast mixer if it's not already running 2670 if (mFastMixer != NULL) { 2671 FastMixerStateQueue *sq = mFastMixer->sq(); 2672 FastMixerState *state = sq->begin(); 2673 if (state->mCommand != FastMixerState::MIX_WRITE && 2674 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2675 if (state->mCommand == FastMixerState::COLD_IDLE) { 2676 int32_t old = android_atomic_inc(&mFastMixerFutex); 2677 if (old == -1) { 2678 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2679 } 2680#ifdef AUDIO_WATCHDOG 2681 if (mAudioWatchdog != 0) { 2682 mAudioWatchdog->resume(); 2683 } 2684#endif 2685 } 2686 state->mCommand = FastMixerState::MIX_WRITE; 2687 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2688 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2689 sq->end(); 2690 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2691 if (kUseFastMixer == FastMixer_Dynamic) { 2692 mNormalSink = mPipeSink; 2693 } 2694 } else { 2695 sq->end(false /*didModify*/); 2696 } 2697 } 2698 return PlaybackThread::threadLoop_write(); 2699} 2700 2701void AudioFlinger::MixerThread::threadLoop_standby() 2702{ 2703 // Idle the fast mixer if it's currently running 2704 if (mFastMixer != NULL) { 2705 FastMixerStateQueue *sq = mFastMixer->sq(); 2706 FastMixerState *state = sq->begin(); 2707 if (!(state->mCommand & FastMixerState::IDLE)) { 2708 state->mCommand = FastMixerState::COLD_IDLE; 2709 state->mColdFutexAddr = &mFastMixerFutex; 2710 state->mColdGen++; 2711 mFastMixerFutex = 0; 2712 sq->end(); 2713 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2714 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2715 if (kUseFastMixer == FastMixer_Dynamic) { 2716 mNormalSink = mOutputSink; 2717 } 2718#ifdef AUDIO_WATCHDOG 2719 if (mAudioWatchdog != 0) { 2720 mAudioWatchdog->pause(); 2721 } 2722#endif 2723 } else { 2724 sq->end(false /*didModify*/); 2725 } 2726 } 2727 PlaybackThread::threadLoop_standby(); 2728} 2729 2730bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2731{ 2732 return false; 2733} 2734 2735bool AudioFlinger::PlaybackThread::shouldStandby_l() 2736{ 2737 return !mStandby; 2738} 2739 2740bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2741{ 2742 Mutex::Autolock _l(mLock); 2743 return waitingAsyncCallback_l(); 2744} 2745 2746// shared by MIXER and DIRECT, overridden by DUPLICATING 2747void AudioFlinger::PlaybackThread::threadLoop_standby() 2748{ 2749 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2750 mOutput->stream->common.standby(&mOutput->stream->common); 2751 if (mUseAsyncWrite != 0) { 2752 // discard any pending drain or write ack by incrementing sequence 2753 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2754 mDrainSequence = (mDrainSequence + 2) & ~1; 2755 ALOG_ASSERT(mCallbackThread != 0); 2756 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2757 mCallbackThread->setDraining(mDrainSequence); 2758 } 2759} 2760 2761void AudioFlinger::MixerThread::threadLoop_mix() 2762{ 2763 // obtain the presentation timestamp of the next output buffer 2764 int64_t pts; 2765 status_t status = INVALID_OPERATION; 2766 2767 if (mNormalSink != 0) { 2768 status = mNormalSink->getNextWriteTimestamp(&pts); 2769 } else { 2770 status = mOutputSink->getNextWriteTimestamp(&pts); 2771 } 2772 2773 if (status != NO_ERROR) { 2774 pts = AudioBufferProvider::kInvalidPTS; 2775 } 2776 2777 // mix buffers... 2778 mAudioMixer->process(pts); 2779 mCurrentWriteLength = mixBufferSize; 2780 // increase sleep time progressively when application underrun condition clears. 2781 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2782 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2783 // such that we would underrun the audio HAL. 2784 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2785 sleepTimeShift--; 2786 } 2787 sleepTime = 0; 2788 standbyTime = systemTime() + standbyDelay; 2789 //TODO: delay standby when effects have a tail 2790} 2791 2792void AudioFlinger::MixerThread::threadLoop_sleepTime() 2793{ 2794 // If no tracks are ready, sleep once for the duration of an output 2795 // buffer size, then write 0s to the output 2796 if (sleepTime == 0) { 2797 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2798 sleepTime = activeSleepTime >> sleepTimeShift; 2799 if (sleepTime < kMinThreadSleepTimeUs) { 2800 sleepTime = kMinThreadSleepTimeUs; 2801 } 2802 // reduce sleep time in case of consecutive application underruns to avoid 2803 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2804 // duration we would end up writing less data than needed by the audio HAL if 2805 // the condition persists. 2806 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2807 sleepTimeShift++; 2808 } 2809 } else { 2810 sleepTime = idleSleepTime; 2811 } 2812 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2813 memset(mMixBuffer, 0, mixBufferSize); 2814 sleepTime = 0; 2815 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2816 "anticipated start"); 2817 } 2818 // TODO add standby time extension fct of effect tail 2819} 2820 2821// prepareTracks_l() must be called with ThreadBase::mLock held 2822AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2823 Vector< sp<Track> > *tracksToRemove) 2824{ 2825 2826 mixer_state mixerStatus = MIXER_IDLE; 2827 // find out which tracks need to be processed 2828 size_t count = mActiveTracks.size(); 2829 size_t mixedTracks = 0; 2830 size_t tracksWithEffect = 0; 2831 // counts only _active_ fast tracks 2832 size_t fastTracks = 0; 2833 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2834 2835 float masterVolume = mMasterVolume; 2836 bool masterMute = mMasterMute; 2837 2838 if (masterMute) { 2839 masterVolume = 0; 2840 } 2841 // Delegate master volume control to effect in output mix effect chain if needed 2842 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2843 if (chain != 0) { 2844 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2845 chain->setVolume_l(&v, &v); 2846 masterVolume = (float)((v + (1 << 23)) >> 24); 2847 chain.clear(); 2848 } 2849 2850 // prepare a new state to push 2851 FastMixerStateQueue *sq = NULL; 2852 FastMixerState *state = NULL; 2853 bool didModify = false; 2854 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2855 if (mFastMixer != NULL) { 2856 sq = mFastMixer->sq(); 2857 state = sq->begin(); 2858 } 2859 2860 for (size_t i=0 ; i<count ; i++) { 2861 const sp<Track> t = mActiveTracks[i].promote(); 2862 if (t == 0) { 2863 continue; 2864 } 2865 2866 // this const just means the local variable doesn't change 2867 Track* const track = t.get(); 2868 2869 // process fast tracks 2870 if (track->isFastTrack()) { 2871 2872 // It's theoretically possible (though unlikely) for a fast track to be created 2873 // and then removed within the same normal mix cycle. This is not a problem, as 2874 // the track never becomes active so it's fast mixer slot is never touched. 2875 // The converse, of removing an (active) track and then creating a new track 2876 // at the identical fast mixer slot within the same normal mix cycle, 2877 // is impossible because the slot isn't marked available until the end of each cycle. 2878 int j = track->mFastIndex; 2879 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2880 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2881 FastTrack *fastTrack = &state->mFastTracks[j]; 2882 2883 // Determine whether the track is currently in underrun condition, 2884 // and whether it had a recent underrun. 2885 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2886 FastTrackUnderruns underruns = ftDump->mUnderruns; 2887 uint32_t recentFull = (underruns.mBitFields.mFull - 2888 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2889 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2890 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2891 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2892 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2893 uint32_t recentUnderruns = recentPartial + recentEmpty; 2894 track->mObservedUnderruns = underruns; 2895 // don't count underruns that occur while stopping or pausing 2896 // or stopped which can occur when flush() is called while active 2897 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2898 recentUnderruns > 0) { 2899 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2900 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2901 } 2902 2903 // This is similar to the state machine for normal tracks, 2904 // with a few modifications for fast tracks. 2905 bool isActive = true; 2906 switch (track->mState) { 2907 case TrackBase::STOPPING_1: 2908 // track stays active in STOPPING_1 state until first underrun 2909 if (recentUnderruns > 0 || track->isTerminated()) { 2910 track->mState = TrackBase::STOPPING_2; 2911 } 2912 break; 2913 case TrackBase::PAUSING: 2914 // ramp down is not yet implemented 2915 track->setPaused(); 2916 break; 2917 case TrackBase::RESUMING: 2918 // ramp up is not yet implemented 2919 track->mState = TrackBase::ACTIVE; 2920 break; 2921 case TrackBase::ACTIVE: 2922 if (recentFull > 0 || recentPartial > 0) { 2923 // track has provided at least some frames recently: reset retry count 2924 track->mRetryCount = kMaxTrackRetries; 2925 } 2926 if (recentUnderruns == 0) { 2927 // no recent underruns: stay active 2928 break; 2929 } 2930 // there has recently been an underrun of some kind 2931 if (track->sharedBuffer() == 0) { 2932 // were any of the recent underruns "empty" (no frames available)? 2933 if (recentEmpty == 0) { 2934 // no, then ignore the partial underruns as they are allowed indefinitely 2935 break; 2936 } 2937 // there has recently been an "empty" underrun: decrement the retry counter 2938 if (--(track->mRetryCount) > 0) { 2939 break; 2940 } 2941 // indicate to client process that the track was disabled because of underrun; 2942 // it will then automatically call start() when data is available 2943 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2944 // remove from active list, but state remains ACTIVE [confusing but true] 2945 isActive = false; 2946 break; 2947 } 2948 // fall through 2949 case TrackBase::STOPPING_2: 2950 case TrackBase::PAUSED: 2951 case TrackBase::STOPPED: 2952 case TrackBase::FLUSHED: // flush() while active 2953 // Check for presentation complete if track is inactive 2954 // We have consumed all the buffers of this track. 2955 // This would be incomplete if we auto-paused on underrun 2956 { 2957 size_t audioHALFrames = 2958 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2959 size_t framesWritten = mBytesWritten / mFrameSize; 2960 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2961 // track stays in active list until presentation is complete 2962 break; 2963 } 2964 } 2965 if (track->isStopping_2()) { 2966 track->mState = TrackBase::STOPPED; 2967 } 2968 if (track->isStopped()) { 2969 // Can't reset directly, as fast mixer is still polling this track 2970 // track->reset(); 2971 // So instead mark this track as needing to be reset after push with ack 2972 resetMask |= 1 << i; 2973 } 2974 isActive = false; 2975 break; 2976 case TrackBase::IDLE: 2977 default: 2978 LOG_FATAL("unexpected track state %d", track->mState); 2979 } 2980 2981 if (isActive) { 2982 // was it previously inactive? 2983 if (!(state->mTrackMask & (1 << j))) { 2984 ExtendedAudioBufferProvider *eabp = track; 2985 VolumeProvider *vp = track; 2986 fastTrack->mBufferProvider = eabp; 2987 fastTrack->mVolumeProvider = vp; 2988 fastTrack->mSampleRate = track->mSampleRate; 2989 fastTrack->mChannelMask = track->mChannelMask; 2990 fastTrack->mGeneration++; 2991 state->mTrackMask |= 1 << j; 2992 didModify = true; 2993 // no acknowledgement required for newly active tracks 2994 } 2995 // cache the combined master volume and stream type volume for fast mixer; this 2996 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2997 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2998 ++fastTracks; 2999 } else { 3000 // was it previously active? 3001 if (state->mTrackMask & (1 << j)) { 3002 fastTrack->mBufferProvider = NULL; 3003 fastTrack->mGeneration++; 3004 state->mTrackMask &= ~(1 << j); 3005 didModify = true; 3006 // If any fast tracks were removed, we must wait for acknowledgement 3007 // because we're about to decrement the last sp<> on those tracks. 3008 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3009 } else { 3010 LOG_FATAL("fast track %d should have been active", j); 3011 } 3012 tracksToRemove->add(track); 3013 // Avoids a misleading display in dumpsys 3014 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3015 } 3016 continue; 3017 } 3018 3019 { // local variable scope to avoid goto warning 3020 3021 audio_track_cblk_t* cblk = track->cblk(); 3022 3023 // The first time a track is added we wait 3024 // for all its buffers to be filled before processing it 3025 int name = track->name(); 3026 // make sure that we have enough frames to mix one full buffer. 3027 // enforce this condition only once to enable draining the buffer in case the client 3028 // app does not call stop() and relies on underrun to stop: 3029 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3030 // during last round 3031 size_t desiredFrames; 3032 uint32_t sr = track->sampleRate(); 3033 if (sr == mSampleRate) { 3034 desiredFrames = mNormalFrameCount; 3035 } else { 3036 // +1 for rounding and +1 for additional sample needed for interpolation 3037 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3038 // add frames already consumed but not yet released by the resampler 3039 // because mAudioTrackServerProxy->framesReady() will include these frames 3040 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3041#if 0 3042 // the minimum track buffer size is normally twice the number of frames necessary 3043 // to fill one buffer and the resampler should not leave more than one buffer worth 3044 // of unreleased frames after each pass, but just in case... 3045 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3046#endif 3047 } 3048 uint32_t minFrames = 1; 3049 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3050 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3051 minFrames = desiredFrames; 3052 } 3053 3054 size_t framesReady = track->framesReady(); 3055 if ((framesReady >= minFrames) && track->isReady() && 3056 !track->isPaused() && !track->isTerminated()) 3057 { 3058 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3059 3060 mixedTracks++; 3061 3062 // track->mainBuffer() != mMixBuffer means there is an effect chain 3063 // connected to the track 3064 chain.clear(); 3065 if (track->mainBuffer() != mMixBuffer) { 3066 chain = getEffectChain_l(track->sessionId()); 3067 // Delegate volume control to effect in track effect chain if needed 3068 if (chain != 0) { 3069 tracksWithEffect++; 3070 } else { 3071 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3072 "session %d", 3073 name, track->sessionId()); 3074 } 3075 } 3076 3077 3078 int param = AudioMixer::VOLUME; 3079 if (track->mFillingUpStatus == Track::FS_FILLED) { 3080 // no ramp for the first volume setting 3081 track->mFillingUpStatus = Track::FS_ACTIVE; 3082 if (track->mState == TrackBase::RESUMING) { 3083 track->mState = TrackBase::ACTIVE; 3084 param = AudioMixer::RAMP_VOLUME; 3085 } 3086 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3087 // FIXME should not make a decision based on mServer 3088 } else if (cblk->mServer != 0) { 3089 // If the track is stopped before the first frame was mixed, 3090 // do not apply ramp 3091 param = AudioMixer::RAMP_VOLUME; 3092 } 3093 3094 // compute volume for this track 3095 uint32_t vl, vr, va; 3096 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3097 vl = vr = va = 0; 3098 if (track->isPausing()) { 3099 track->setPaused(); 3100 } 3101 } else { 3102 3103 // read original volumes with volume control 3104 float typeVolume = mStreamTypes[track->streamType()].volume; 3105 float v = masterVolume * typeVolume; 3106 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3107 uint32_t vlr = proxy->getVolumeLR(); 3108 vl = vlr & 0xFFFF; 3109 vr = vlr >> 16; 3110 // track volumes come from shared memory, so can't be trusted and must be clamped 3111 if (vl > MAX_GAIN_INT) { 3112 ALOGV("Track left volume out of range: %04X", vl); 3113 vl = MAX_GAIN_INT; 3114 } 3115 if (vr > MAX_GAIN_INT) { 3116 ALOGV("Track right volume out of range: %04X", vr); 3117 vr = MAX_GAIN_INT; 3118 } 3119 // now apply the master volume and stream type volume 3120 vl = (uint32_t)(v * vl) << 12; 3121 vr = (uint32_t)(v * vr) << 12; 3122 // assuming master volume and stream type volume each go up to 1.0, 3123 // vl and vr are now in 8.24 format 3124 3125 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3126 // send level comes from shared memory and so may be corrupt 3127 if (sendLevel > MAX_GAIN_INT) { 3128 ALOGV("Track send level out of range: %04X", sendLevel); 3129 sendLevel = MAX_GAIN_INT; 3130 } 3131 va = (uint32_t)(v * sendLevel); 3132 } 3133 3134 // Delegate volume control to effect in track effect chain if needed 3135 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3136 // Do not ramp volume if volume is controlled by effect 3137 param = AudioMixer::VOLUME; 3138 track->mHasVolumeController = true; 3139 } else { 3140 // force no volume ramp when volume controller was just disabled or removed 3141 // from effect chain to avoid volume spike 3142 if (track->mHasVolumeController) { 3143 param = AudioMixer::VOLUME; 3144 } 3145 track->mHasVolumeController = false; 3146 } 3147 3148 // Convert volumes from 8.24 to 4.12 format 3149 // This additional clamping is needed in case chain->setVolume_l() overshot 3150 vl = (vl + (1 << 11)) >> 12; 3151 if (vl > MAX_GAIN_INT) { 3152 vl = MAX_GAIN_INT; 3153 } 3154 vr = (vr + (1 << 11)) >> 12; 3155 if (vr > MAX_GAIN_INT) { 3156 vr = MAX_GAIN_INT; 3157 } 3158 3159 if (va > MAX_GAIN_INT) { 3160 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3161 } 3162 3163 // XXX: these things DON'T need to be done each time 3164 mAudioMixer->setBufferProvider(name, track); 3165 mAudioMixer->enable(name); 3166 3167 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3168 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3169 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3170 mAudioMixer->setParameter( 3171 name, 3172 AudioMixer::TRACK, 3173 AudioMixer::FORMAT, (void *)track->format()); 3174 mAudioMixer->setParameter( 3175 name, 3176 AudioMixer::TRACK, 3177 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3178 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3179 uint32_t maxSampleRate = mSampleRate * 2; 3180 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3181 if (reqSampleRate == 0) { 3182 reqSampleRate = mSampleRate; 3183 } else if (reqSampleRate > maxSampleRate) { 3184 reqSampleRate = maxSampleRate; 3185 } 3186 mAudioMixer->setParameter( 3187 name, 3188 AudioMixer::RESAMPLE, 3189 AudioMixer::SAMPLE_RATE, 3190 (void *)reqSampleRate); 3191 mAudioMixer->setParameter( 3192 name, 3193 AudioMixer::TRACK, 3194 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3195 mAudioMixer->setParameter( 3196 name, 3197 AudioMixer::TRACK, 3198 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3199 3200 // reset retry count 3201 track->mRetryCount = kMaxTrackRetries; 3202 3203 // If one track is ready, set the mixer ready if: 3204 // - the mixer was not ready during previous round OR 3205 // - no other track is not ready 3206 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3207 mixerStatus != MIXER_TRACKS_ENABLED) { 3208 mixerStatus = MIXER_TRACKS_READY; 3209 } 3210 } else { 3211 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3212 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3213 } 3214 // clear effect chain input buffer if an active track underruns to avoid sending 3215 // previous audio buffer again to effects 3216 chain = getEffectChain_l(track->sessionId()); 3217 if (chain != 0) { 3218 chain->clearInputBuffer(); 3219 } 3220 3221 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3222 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3223 track->isStopped() || track->isPaused()) { 3224 // We have consumed all the buffers of this track. 3225 // Remove it from the list of active tracks. 3226 // TODO: use actual buffer filling status instead of latency when available from 3227 // audio HAL 3228 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3229 size_t framesWritten = mBytesWritten / mFrameSize; 3230 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3231 if (track->isStopped()) { 3232 track->reset(); 3233 } 3234 tracksToRemove->add(track); 3235 } 3236 } else { 3237 // No buffers for this track. Give it a few chances to 3238 // fill a buffer, then remove it from active list. 3239 if (--(track->mRetryCount) <= 0) { 3240 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3241 tracksToRemove->add(track); 3242 // indicate to client process that the track was disabled because of underrun; 3243 // it will then automatically call start() when data is available 3244 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3245 // If one track is not ready, mark the mixer also not ready if: 3246 // - the mixer was ready during previous round OR 3247 // - no other track is ready 3248 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3249 mixerStatus != MIXER_TRACKS_READY) { 3250 mixerStatus = MIXER_TRACKS_ENABLED; 3251 } 3252 } 3253 mAudioMixer->disable(name); 3254 } 3255 3256 } // local variable scope to avoid goto warning 3257track_is_ready: ; 3258 3259 } 3260 3261 // Push the new FastMixer state if necessary 3262 bool pauseAudioWatchdog = false; 3263 if (didModify) { 3264 state->mFastTracksGen++; 3265 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3266 if (kUseFastMixer == FastMixer_Dynamic && 3267 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3268 state->mCommand = FastMixerState::COLD_IDLE; 3269 state->mColdFutexAddr = &mFastMixerFutex; 3270 state->mColdGen++; 3271 mFastMixerFutex = 0; 3272 if (kUseFastMixer == FastMixer_Dynamic) { 3273 mNormalSink = mOutputSink; 3274 } 3275 // If we go into cold idle, need to wait for acknowledgement 3276 // so that fast mixer stops doing I/O. 3277 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3278 pauseAudioWatchdog = true; 3279 } 3280 } 3281 if (sq != NULL) { 3282 sq->end(didModify); 3283 sq->push(block); 3284 } 3285#ifdef AUDIO_WATCHDOG 3286 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3287 mAudioWatchdog->pause(); 3288 } 3289#endif 3290 3291 // Now perform the deferred reset on fast tracks that have stopped 3292 while (resetMask != 0) { 3293 size_t i = __builtin_ctz(resetMask); 3294 ALOG_ASSERT(i < count); 3295 resetMask &= ~(1 << i); 3296 sp<Track> t = mActiveTracks[i].promote(); 3297 if (t == 0) { 3298 continue; 3299 } 3300 Track* track = t.get(); 3301 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3302 track->reset(); 3303 } 3304 3305 // remove all the tracks that need to be... 3306 removeTracks_l(*tracksToRemove); 3307 3308 // mix buffer must be cleared if all tracks are connected to an 3309 // effect chain as in this case the mixer will not write to 3310 // mix buffer and track effects will accumulate into it 3311 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3312 (mixedTracks == 0 && fastTracks > 0))) { 3313 // FIXME as a performance optimization, should remember previous zero status 3314 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3315 } 3316 3317 // if any fast tracks, then status is ready 3318 mMixerStatusIgnoringFastTracks = mixerStatus; 3319 if (fastTracks > 0) { 3320 mixerStatus = MIXER_TRACKS_READY; 3321 } 3322 return mixerStatus; 3323} 3324 3325// getTrackName_l() must be called with ThreadBase::mLock held 3326int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3327{ 3328 return mAudioMixer->getTrackName(channelMask, sessionId); 3329} 3330 3331// deleteTrackName_l() must be called with ThreadBase::mLock held 3332void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3333{ 3334 ALOGV("remove track (%d) and delete from mixer", name); 3335 mAudioMixer->deleteTrackName(name); 3336} 3337 3338// checkForNewParameters_l() must be called with ThreadBase::mLock held 3339bool AudioFlinger::MixerThread::checkForNewParameters_l() 3340{ 3341 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3342 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3343 bool reconfig = false; 3344 3345 while (!mNewParameters.isEmpty()) { 3346 3347 if (mFastMixer != NULL) { 3348 FastMixerStateQueue *sq = mFastMixer->sq(); 3349 FastMixerState *state = sq->begin(); 3350 if (!(state->mCommand & FastMixerState::IDLE)) { 3351 previousCommand = state->mCommand; 3352 state->mCommand = FastMixerState::HOT_IDLE; 3353 sq->end(); 3354 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3355 } else { 3356 sq->end(false /*didModify*/); 3357 } 3358 } 3359 3360 status_t status = NO_ERROR; 3361 String8 keyValuePair = mNewParameters[0]; 3362 AudioParameter param = AudioParameter(keyValuePair); 3363 int value; 3364 3365 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3366 reconfig = true; 3367 } 3368 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3369 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3370 status = BAD_VALUE; 3371 } else { 3372 // no need to save value, since it's constant 3373 reconfig = true; 3374 } 3375 } 3376 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3377 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3378 status = BAD_VALUE; 3379 } else { 3380 // no need to save value, since it's constant 3381 reconfig = true; 3382 } 3383 } 3384 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3385 // do not accept frame count changes if tracks are open as the track buffer 3386 // size depends on frame count and correct behavior would not be guaranteed 3387 // if frame count is changed after track creation 3388 if (!mTracks.isEmpty()) { 3389 status = INVALID_OPERATION; 3390 } else { 3391 reconfig = true; 3392 } 3393 } 3394 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3395#ifdef ADD_BATTERY_DATA 3396 // when changing the audio output device, call addBatteryData to notify 3397 // the change 3398 if (mOutDevice != value) { 3399 uint32_t params = 0; 3400 // check whether speaker is on 3401 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3402 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3403 } 3404 3405 audio_devices_t deviceWithoutSpeaker 3406 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3407 // check if any other device (except speaker) is on 3408 if (value & deviceWithoutSpeaker ) { 3409 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3410 } 3411 3412 if (params != 0) { 3413 addBatteryData(params); 3414 } 3415 } 3416#endif 3417 3418 // forward device change to effects that have requested to be 3419 // aware of attached audio device. 3420 if (value != AUDIO_DEVICE_NONE) { 3421 mOutDevice = value; 3422 for (size_t i = 0; i < mEffectChains.size(); i++) { 3423 mEffectChains[i]->setDevice_l(mOutDevice); 3424 } 3425 } 3426 } 3427 3428 if (status == NO_ERROR) { 3429 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3430 keyValuePair.string()); 3431 if (!mStandby && status == INVALID_OPERATION) { 3432 mOutput->stream->common.standby(&mOutput->stream->common); 3433 mStandby = true; 3434 mBytesWritten = 0; 3435 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3436 keyValuePair.string()); 3437 } 3438 if (status == NO_ERROR && reconfig) { 3439 readOutputParameters(); 3440 delete mAudioMixer; 3441 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3442 for (size_t i = 0; i < mTracks.size() ; i++) { 3443 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3444 if (name < 0) { 3445 break; 3446 } 3447 mTracks[i]->mName = name; 3448 } 3449 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3450 } 3451 } 3452 3453 mNewParameters.removeAt(0); 3454 3455 mParamStatus = status; 3456 mParamCond.signal(); 3457 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3458 // already timed out waiting for the status and will never signal the condition. 3459 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3460 } 3461 3462 if (!(previousCommand & FastMixerState::IDLE)) { 3463 ALOG_ASSERT(mFastMixer != NULL); 3464 FastMixerStateQueue *sq = mFastMixer->sq(); 3465 FastMixerState *state = sq->begin(); 3466 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3467 state->mCommand = previousCommand; 3468 sq->end(); 3469 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3470 } 3471 3472 return reconfig; 3473} 3474 3475 3476void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3477{ 3478 const size_t SIZE = 256; 3479 char buffer[SIZE]; 3480 String8 result; 3481 3482 PlaybackThread::dumpInternals(fd, args); 3483 3484 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3485 result.append(buffer); 3486 write(fd, result.string(), result.size()); 3487 3488 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3489 const FastMixerDumpState copy(mFastMixerDumpState); 3490 copy.dump(fd); 3491 3492#ifdef STATE_QUEUE_DUMP 3493 // Similar for state queue 3494 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3495 observerCopy.dump(fd); 3496 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3497 mutatorCopy.dump(fd); 3498#endif 3499 3500#ifdef TEE_SINK 3501 // Write the tee output to a .wav file 3502 dumpTee(fd, mTeeSource, mId); 3503#endif 3504 3505#ifdef AUDIO_WATCHDOG 3506 if (mAudioWatchdog != 0) { 3507 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3508 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3509 wdCopy.dump(fd); 3510 } 3511#endif 3512} 3513 3514uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3515{ 3516 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3517} 3518 3519uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3520{ 3521 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3522} 3523 3524void AudioFlinger::MixerThread::cacheParameters_l() 3525{ 3526 PlaybackThread::cacheParameters_l(); 3527 3528 // FIXME: Relaxed timing because of a certain device that can't meet latency 3529 // Should be reduced to 2x after the vendor fixes the driver issue 3530 // increase threshold again due to low power audio mode. The way this warning 3531 // threshold is calculated and its usefulness should be reconsidered anyway. 3532 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3533} 3534 3535// ---------------------------------------------------------------------------- 3536 3537AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3538 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3539 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3540 // mLeftVolFloat, mRightVolFloat 3541{ 3542} 3543 3544AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3545 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3546 ThreadBase::type_t type) 3547 : PlaybackThread(audioFlinger, output, id, device, type) 3548 // mLeftVolFloat, mRightVolFloat 3549{ 3550} 3551 3552AudioFlinger::DirectOutputThread::~DirectOutputThread() 3553{ 3554} 3555 3556void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3557{ 3558 audio_track_cblk_t* cblk = track->cblk(); 3559 float left, right; 3560 3561 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3562 left = right = 0; 3563 } else { 3564 float typeVolume = mStreamTypes[track->streamType()].volume; 3565 float v = mMasterVolume * typeVolume; 3566 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3567 uint32_t vlr = proxy->getVolumeLR(); 3568 float v_clamped = v * (vlr & 0xFFFF); 3569 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3570 left = v_clamped/MAX_GAIN; 3571 v_clamped = v * (vlr >> 16); 3572 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3573 right = v_clamped/MAX_GAIN; 3574 } 3575 3576 if (lastTrack) { 3577 if (left != mLeftVolFloat || right != mRightVolFloat) { 3578 mLeftVolFloat = left; 3579 mRightVolFloat = right; 3580 3581 // Convert volumes from float to 8.24 3582 uint32_t vl = (uint32_t)(left * (1 << 24)); 3583 uint32_t vr = (uint32_t)(right * (1 << 24)); 3584 3585 // Delegate volume control to effect in track effect chain if needed 3586 // only one effect chain can be present on DirectOutputThread, so if 3587 // there is one, the track is connected to it 3588 if (!mEffectChains.isEmpty()) { 3589 mEffectChains[0]->setVolume_l(&vl, &vr); 3590 left = (float)vl / (1 << 24); 3591 right = (float)vr / (1 << 24); 3592 } 3593 if (mOutput->stream->set_volume) { 3594 mOutput->stream->set_volume(mOutput->stream, left, right); 3595 } 3596 } 3597 } 3598} 3599 3600 3601AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3602 Vector< sp<Track> > *tracksToRemove 3603) 3604{ 3605 size_t count = mActiveTracks.size(); 3606 mixer_state mixerStatus = MIXER_IDLE; 3607 3608 // find out which tracks need to be processed 3609 for (size_t i = 0; i < count; i++) { 3610 sp<Track> t = mActiveTracks[i].promote(); 3611 // The track died recently 3612 if (t == 0) { 3613 continue; 3614 } 3615 3616 Track* const track = t.get(); 3617 audio_track_cblk_t* cblk = track->cblk(); 3618 // Only consider last track started for volume and mixer state control. 3619 // In theory an older track could underrun and restart after the new one starts 3620 // but as we only care about the transition phase between two tracks on a 3621 // direct output, it is not a problem to ignore the underrun case. 3622 sp<Track> l = mLatestActiveTrack.promote(); 3623 bool last = l.get() == track; 3624 3625 // The first time a track is added we wait 3626 // for all its buffers to be filled before processing it 3627 uint32_t minFrames; 3628 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3629 minFrames = mNormalFrameCount; 3630 } else { 3631 minFrames = 1; 3632 } 3633 3634 if ((track->framesReady() >= minFrames) && track->isReady() && 3635 !track->isPaused() && !track->isTerminated()) 3636 { 3637 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3638 3639 if (track->mFillingUpStatus == Track::FS_FILLED) { 3640 track->mFillingUpStatus = Track::FS_ACTIVE; 3641 // make sure processVolume_l() will apply new volume even if 0 3642 mLeftVolFloat = mRightVolFloat = -1.0; 3643 if (track->mState == TrackBase::RESUMING) { 3644 track->mState = TrackBase::ACTIVE; 3645 } 3646 } 3647 3648 // compute volume for this track 3649 processVolume_l(track, last); 3650 if (last) { 3651 // reset retry count 3652 track->mRetryCount = kMaxTrackRetriesDirect; 3653 mActiveTrack = t; 3654 mixerStatus = MIXER_TRACKS_READY; 3655 } 3656 } else { 3657 // clear effect chain input buffer if the last active track started underruns 3658 // to avoid sending previous audio buffer again to effects 3659 if (!mEffectChains.isEmpty() && last) { 3660 mEffectChains[0]->clearInputBuffer(); 3661 } 3662 3663 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3664 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3665 track->isStopped() || track->isPaused()) { 3666 // We have consumed all the buffers of this track. 3667 // Remove it from the list of active tracks. 3668 // TODO: implement behavior for compressed audio 3669 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3670 size_t framesWritten = mBytesWritten / mFrameSize; 3671 if (mStandby || !last || 3672 track->presentationComplete(framesWritten, audioHALFrames)) { 3673 if (track->isStopped()) { 3674 track->reset(); 3675 } 3676 tracksToRemove->add(track); 3677 } 3678 } else { 3679 // No buffers for this track. Give it a few chances to 3680 // fill a buffer, then remove it from active list. 3681 // Only consider last track started for mixer state control 3682 if (--(track->mRetryCount) <= 0) { 3683 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3684 tracksToRemove->add(track); 3685 // indicate to client process that the track was disabled because of underrun; 3686 // it will then automatically call start() when data is available 3687 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3688 } else if (last) { 3689 mixerStatus = MIXER_TRACKS_ENABLED; 3690 } 3691 } 3692 } 3693 } 3694 3695 // remove all the tracks that need to be... 3696 removeTracks_l(*tracksToRemove); 3697 3698 return mixerStatus; 3699} 3700 3701void AudioFlinger::DirectOutputThread::threadLoop_mix() 3702{ 3703 size_t frameCount = mFrameCount; 3704 int8_t *curBuf = (int8_t *)mMixBuffer; 3705 // output audio to hardware 3706 while (frameCount) { 3707 AudioBufferProvider::Buffer buffer; 3708 buffer.frameCount = frameCount; 3709 mActiveTrack->getNextBuffer(&buffer); 3710 if (buffer.raw == NULL) { 3711 memset(curBuf, 0, frameCount * mFrameSize); 3712 break; 3713 } 3714 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3715 frameCount -= buffer.frameCount; 3716 curBuf += buffer.frameCount * mFrameSize; 3717 mActiveTrack->releaseBuffer(&buffer); 3718 } 3719 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3720 sleepTime = 0; 3721 standbyTime = systemTime() + standbyDelay; 3722 mActiveTrack.clear(); 3723} 3724 3725void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3726{ 3727 if (sleepTime == 0) { 3728 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3729 sleepTime = activeSleepTime; 3730 } else { 3731 sleepTime = idleSleepTime; 3732 } 3733 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3734 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3735 sleepTime = 0; 3736 } 3737} 3738 3739// getTrackName_l() must be called with ThreadBase::mLock held 3740int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3741 int sessionId __unused) 3742{ 3743 return 0; 3744} 3745 3746// deleteTrackName_l() must be called with ThreadBase::mLock held 3747void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3748{ 3749} 3750 3751// checkForNewParameters_l() must be called with ThreadBase::mLock held 3752bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3753{ 3754 bool reconfig = false; 3755 3756 while (!mNewParameters.isEmpty()) { 3757 status_t status = NO_ERROR; 3758 String8 keyValuePair = mNewParameters[0]; 3759 AudioParameter param = AudioParameter(keyValuePair); 3760 int value; 3761 3762 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3763 // do not accept frame count changes if tracks are open as the track buffer 3764 // size depends on frame count and correct behavior would not be garantied 3765 // if frame count is changed after track creation 3766 if (!mTracks.isEmpty()) { 3767 status = INVALID_OPERATION; 3768 } else { 3769 reconfig = true; 3770 } 3771 } 3772 if (status == NO_ERROR) { 3773 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3774 keyValuePair.string()); 3775 if (!mStandby && status == INVALID_OPERATION) { 3776 mOutput->stream->common.standby(&mOutput->stream->common); 3777 mStandby = true; 3778 mBytesWritten = 0; 3779 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3780 keyValuePair.string()); 3781 } 3782 if (status == NO_ERROR && reconfig) { 3783 readOutputParameters(); 3784 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3785 } 3786 } 3787 3788 mNewParameters.removeAt(0); 3789 3790 mParamStatus = status; 3791 mParamCond.signal(); 3792 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3793 // already timed out waiting for the status and will never signal the condition. 3794 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3795 } 3796 return reconfig; 3797} 3798 3799uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3800{ 3801 uint32_t time; 3802 if (audio_is_linear_pcm(mFormat)) { 3803 time = PlaybackThread::activeSleepTimeUs(); 3804 } else { 3805 time = 10000; 3806 } 3807 return time; 3808} 3809 3810uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3811{ 3812 uint32_t time; 3813 if (audio_is_linear_pcm(mFormat)) { 3814 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3815 } else { 3816 time = 10000; 3817 } 3818 return time; 3819} 3820 3821uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3822{ 3823 uint32_t time; 3824 if (audio_is_linear_pcm(mFormat)) { 3825 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3826 } else { 3827 time = 10000; 3828 } 3829 return time; 3830} 3831 3832void AudioFlinger::DirectOutputThread::cacheParameters_l() 3833{ 3834 PlaybackThread::cacheParameters_l(); 3835 3836 // use shorter standby delay as on normal output to release 3837 // hardware resources as soon as possible 3838 if (audio_is_linear_pcm(mFormat)) { 3839 standbyDelay = microseconds(activeSleepTime*2); 3840 } else { 3841 standbyDelay = kOffloadStandbyDelayNs; 3842 } 3843} 3844 3845// ---------------------------------------------------------------------------- 3846 3847AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3848 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3849 : Thread(false /*canCallJava*/), 3850 mPlaybackThread(playbackThread), 3851 mWriteAckSequence(0), 3852 mDrainSequence(0) 3853{ 3854} 3855 3856AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3857{ 3858} 3859 3860void AudioFlinger::AsyncCallbackThread::onFirstRef() 3861{ 3862 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3863} 3864 3865bool AudioFlinger::AsyncCallbackThread::threadLoop() 3866{ 3867 while (!exitPending()) { 3868 uint32_t writeAckSequence; 3869 uint32_t drainSequence; 3870 3871 { 3872 Mutex::Autolock _l(mLock); 3873 while (!((mWriteAckSequence & 1) || 3874 (mDrainSequence & 1) || 3875 exitPending())) { 3876 mWaitWorkCV.wait(mLock); 3877 } 3878 3879 if (exitPending()) { 3880 break; 3881 } 3882 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3883 mWriteAckSequence, mDrainSequence); 3884 writeAckSequence = mWriteAckSequence; 3885 mWriteAckSequence &= ~1; 3886 drainSequence = mDrainSequence; 3887 mDrainSequence &= ~1; 3888 } 3889 { 3890 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3891 if (playbackThread != 0) { 3892 if (writeAckSequence & 1) { 3893 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3894 } 3895 if (drainSequence & 1) { 3896 playbackThread->resetDraining(drainSequence >> 1); 3897 } 3898 } 3899 } 3900 } 3901 return false; 3902} 3903 3904void AudioFlinger::AsyncCallbackThread::exit() 3905{ 3906 ALOGV("AsyncCallbackThread::exit"); 3907 Mutex::Autolock _l(mLock); 3908 requestExit(); 3909 mWaitWorkCV.broadcast(); 3910} 3911 3912void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3913{ 3914 Mutex::Autolock _l(mLock); 3915 // bit 0 is cleared 3916 mWriteAckSequence = sequence << 1; 3917} 3918 3919void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3920{ 3921 Mutex::Autolock _l(mLock); 3922 // ignore unexpected callbacks 3923 if (mWriteAckSequence & 2) { 3924 mWriteAckSequence |= 1; 3925 mWaitWorkCV.signal(); 3926 } 3927} 3928 3929void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3930{ 3931 Mutex::Autolock _l(mLock); 3932 // bit 0 is cleared 3933 mDrainSequence = sequence << 1; 3934} 3935 3936void AudioFlinger::AsyncCallbackThread::resetDraining() 3937{ 3938 Mutex::Autolock _l(mLock); 3939 // ignore unexpected callbacks 3940 if (mDrainSequence & 2) { 3941 mDrainSequence |= 1; 3942 mWaitWorkCV.signal(); 3943 } 3944} 3945 3946 3947// ---------------------------------------------------------------------------- 3948AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3949 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3950 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3951 mHwPaused(false), 3952 mFlushPending(false), 3953 mPausedBytesRemaining(0) 3954{ 3955 //FIXME: mStandby should be set to true by ThreadBase constructor 3956 mStandby = true; 3957} 3958 3959void AudioFlinger::OffloadThread::threadLoop_exit() 3960{ 3961 if (mFlushPending || mHwPaused) { 3962 // If a flush is pending or track was paused, just discard buffered data 3963 flushHw_l(); 3964 } else { 3965 mMixerStatus = MIXER_DRAIN_ALL; 3966 threadLoop_drain(); 3967 } 3968 mCallbackThread->exit(); 3969 PlaybackThread::threadLoop_exit(); 3970} 3971 3972AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3973 Vector< sp<Track> > *tracksToRemove 3974) 3975{ 3976 size_t count = mActiveTracks.size(); 3977 3978 mixer_state mixerStatus = MIXER_IDLE; 3979 bool doHwPause = false; 3980 bool doHwResume = false; 3981 3982 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3983 3984 // find out which tracks need to be processed 3985 for (size_t i = 0; i < count; i++) { 3986 sp<Track> t = mActiveTracks[i].promote(); 3987 // The track died recently 3988 if (t == 0) { 3989 continue; 3990 } 3991 Track* const track = t.get(); 3992 audio_track_cblk_t* cblk = track->cblk(); 3993 // Only consider last track started for volume and mixer state control. 3994 // In theory an older track could underrun and restart after the new one starts 3995 // but as we only care about the transition phase between two tracks on a 3996 // direct output, it is not a problem to ignore the underrun case. 3997 sp<Track> l = mLatestActiveTrack.promote(); 3998 bool last = l.get() == track; 3999 4000 if (track->isInvalid()) { 4001 ALOGW("An invalidated track shouldn't be in active list"); 4002 tracksToRemove->add(track); 4003 continue; 4004 } 4005 4006 if (track->mState == TrackBase::IDLE) { 4007 ALOGW("An idle track shouldn't be in active list"); 4008 continue; 4009 } 4010 4011 if (track->isPausing()) { 4012 track->setPaused(); 4013 if (last) { 4014 if (!mHwPaused) { 4015 doHwPause = true; 4016 mHwPaused = true; 4017 } 4018 // If we were part way through writing the mixbuffer to 4019 // the HAL we must save this until we resume 4020 // BUG - this will be wrong if a different track is made active, 4021 // in that case we want to discard the pending data in the 4022 // mixbuffer and tell the client to present it again when the 4023 // track is resumed 4024 mPausedWriteLength = mCurrentWriteLength; 4025 mPausedBytesRemaining = mBytesRemaining; 4026 mBytesRemaining = 0; // stop writing 4027 } 4028 tracksToRemove->add(track); 4029 } else if (track->isFlushPending()) { 4030 track->flushAck(); 4031 if (last) { 4032 mFlushPending = true; 4033 } 4034 } else if (track->framesReady() && track->isReady() && 4035 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4036 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4037 if (track->mFillingUpStatus == Track::FS_FILLED) { 4038 track->mFillingUpStatus = Track::FS_ACTIVE; 4039 // make sure processVolume_l() will apply new volume even if 0 4040 mLeftVolFloat = mRightVolFloat = -1.0; 4041 if (track->mState == TrackBase::RESUMING) { 4042 track->mState = TrackBase::ACTIVE; 4043 if (last) { 4044 if (mPausedBytesRemaining) { 4045 // Need to continue write that was interrupted 4046 mCurrentWriteLength = mPausedWriteLength; 4047 mBytesRemaining = mPausedBytesRemaining; 4048 mPausedBytesRemaining = 0; 4049 } 4050 if (mHwPaused) { 4051 doHwResume = true; 4052 mHwPaused = false; 4053 // threadLoop_mix() will handle the case that we need to 4054 // resume an interrupted write 4055 } 4056 // enable write to audio HAL 4057 sleepTime = 0; 4058 } 4059 } 4060 } 4061 4062 if (last) { 4063 sp<Track> previousTrack = mPreviousTrack.promote(); 4064 if (previousTrack != 0) { 4065 if (track != previousTrack.get()) { 4066 // Flush any data still being written from last track 4067 mBytesRemaining = 0; 4068 if (mPausedBytesRemaining) { 4069 // Last track was paused so we also need to flush saved 4070 // mixbuffer state and invalidate track so that it will 4071 // re-submit that unwritten data when it is next resumed 4072 mPausedBytesRemaining = 0; 4073 // Invalidate is a bit drastic - would be more efficient 4074 // to have a flag to tell client that some of the 4075 // previously written data was lost 4076 previousTrack->invalidate(); 4077 } 4078 // flush data already sent to the DSP if changing audio session as audio 4079 // comes from a different source. Also invalidate previous track to force a 4080 // seek when resuming. 4081 if (previousTrack->sessionId() != track->sessionId()) { 4082 previousTrack->invalidate(); 4083 mFlushPending = true; 4084 } 4085 } 4086 } 4087 mPreviousTrack = track; 4088 // reset retry count 4089 track->mRetryCount = kMaxTrackRetriesOffload; 4090 mActiveTrack = t; 4091 mixerStatus = MIXER_TRACKS_READY; 4092 } 4093 } else { 4094 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4095 if (track->isStopping_1()) { 4096 // Hardware buffer can hold a large amount of audio so we must 4097 // wait for all current track's data to drain before we say 4098 // that the track is stopped. 4099 if (mBytesRemaining == 0) { 4100 // Only start draining when all data in mixbuffer 4101 // has been written 4102 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4103 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4104 // do not drain if no data was ever sent to HAL (mStandby == true) 4105 if (last && !mStandby) { 4106 // do not modify drain sequence if we are already draining. This happens 4107 // when resuming from pause after drain. 4108 if ((mDrainSequence & 1) == 0) { 4109 sleepTime = 0; 4110 standbyTime = systemTime() + standbyDelay; 4111 mixerStatus = MIXER_DRAIN_TRACK; 4112 mDrainSequence += 2; 4113 } 4114 if (mHwPaused) { 4115 // It is possible to move from PAUSED to STOPPING_1 without 4116 // a resume so we must ensure hardware is running 4117 doHwResume = true; 4118 mHwPaused = false; 4119 } 4120 } 4121 } 4122 } else if (track->isStopping_2()) { 4123 // Drain has completed or we are in standby, signal presentation complete 4124 if (!(mDrainSequence & 1) || !last || mStandby) { 4125 track->mState = TrackBase::STOPPED; 4126 size_t audioHALFrames = 4127 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4128 size_t framesWritten = 4129 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4130 track->presentationComplete(framesWritten, audioHALFrames); 4131 track->reset(); 4132 tracksToRemove->add(track); 4133 } 4134 } else { 4135 // No buffers for this track. Give it a few chances to 4136 // fill a buffer, then remove it from active list. 4137 if (--(track->mRetryCount) <= 0) { 4138 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4139 track->name()); 4140 tracksToRemove->add(track); 4141 // indicate to client process that the track was disabled because of underrun; 4142 // it will then automatically call start() when data is available 4143 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4144 } else if (last){ 4145 mixerStatus = MIXER_TRACKS_ENABLED; 4146 } 4147 } 4148 } 4149 // compute volume for this track 4150 processVolume_l(track, last); 4151 } 4152 4153 // make sure the pause/flush/resume sequence is executed in the right order. 4154 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4155 // before flush and then resume HW. This can happen in case of pause/flush/resume 4156 // if resume is received before pause is executed. 4157 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4158 mOutput->stream->pause(mOutput->stream); 4159 } 4160 if (mFlushPending) { 4161 flushHw_l(); 4162 mFlushPending = false; 4163 } 4164 if (!mStandby && doHwResume) { 4165 mOutput->stream->resume(mOutput->stream); 4166 } 4167 4168 // remove all the tracks that need to be... 4169 removeTracks_l(*tracksToRemove); 4170 4171 return mixerStatus; 4172} 4173 4174// must be called with thread mutex locked 4175bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4176{ 4177 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4178 mWriteAckSequence, mDrainSequence); 4179 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4180 return true; 4181 } 4182 return false; 4183} 4184 4185// must be called with thread mutex locked 4186bool AudioFlinger::OffloadThread::shouldStandby_l() 4187{ 4188 bool trackPaused = false; 4189 4190 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4191 // after a timeout and we will enter standby then. 4192 if (mTracks.size() > 0) { 4193 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4194 } 4195 4196 return !mStandby && !trackPaused; 4197} 4198 4199 4200bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4201{ 4202 Mutex::Autolock _l(mLock); 4203 return waitingAsyncCallback_l(); 4204} 4205 4206void AudioFlinger::OffloadThread::flushHw_l() 4207{ 4208 mOutput->stream->flush(mOutput->stream); 4209 // Flush anything still waiting in the mixbuffer 4210 mCurrentWriteLength = 0; 4211 mBytesRemaining = 0; 4212 mPausedWriteLength = 0; 4213 mPausedBytesRemaining = 0; 4214 mHwPaused = false; 4215 4216 if (mUseAsyncWrite) { 4217 // discard any pending drain or write ack by incrementing sequence 4218 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4219 mDrainSequence = (mDrainSequence + 2) & ~1; 4220 ALOG_ASSERT(mCallbackThread != 0); 4221 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4222 mCallbackThread->setDraining(mDrainSequence); 4223 } 4224} 4225 4226// ---------------------------------------------------------------------------- 4227 4228AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4229 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4230 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4231 DUPLICATING), 4232 mWaitTimeMs(UINT_MAX) 4233{ 4234 addOutputTrack(mainThread); 4235} 4236 4237AudioFlinger::DuplicatingThread::~DuplicatingThread() 4238{ 4239 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4240 mOutputTracks[i]->destroy(); 4241 } 4242} 4243 4244void AudioFlinger::DuplicatingThread::threadLoop_mix() 4245{ 4246 // mix buffers... 4247 if (outputsReady(outputTracks)) { 4248 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4249 } else { 4250 memset(mMixBuffer, 0, mixBufferSize); 4251 } 4252 sleepTime = 0; 4253 writeFrames = mNormalFrameCount; 4254 mCurrentWriteLength = mixBufferSize; 4255 standbyTime = systemTime() + standbyDelay; 4256} 4257 4258void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4259{ 4260 if (sleepTime == 0) { 4261 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4262 sleepTime = activeSleepTime; 4263 } else { 4264 sleepTime = idleSleepTime; 4265 } 4266 } else if (mBytesWritten != 0) { 4267 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4268 writeFrames = mNormalFrameCount; 4269 memset(mMixBuffer, 0, mixBufferSize); 4270 } else { 4271 // flush remaining overflow buffers in output tracks 4272 writeFrames = 0; 4273 } 4274 sleepTime = 0; 4275 } 4276} 4277 4278ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4279{ 4280 for (size_t i = 0; i < outputTracks.size(); i++) { 4281 outputTracks[i]->write(mMixBuffer, writeFrames); 4282 } 4283 mStandby = false; 4284 return (ssize_t)mixBufferSize; 4285} 4286 4287void AudioFlinger::DuplicatingThread::threadLoop_standby() 4288{ 4289 // DuplicatingThread implements standby by stopping all tracks 4290 for (size_t i = 0; i < outputTracks.size(); i++) { 4291 outputTracks[i]->stop(); 4292 } 4293} 4294 4295void AudioFlinger::DuplicatingThread::saveOutputTracks() 4296{ 4297 outputTracks = mOutputTracks; 4298} 4299 4300void AudioFlinger::DuplicatingThread::clearOutputTracks() 4301{ 4302 outputTracks.clear(); 4303} 4304 4305void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4306{ 4307 Mutex::Autolock _l(mLock); 4308 // FIXME explain this formula 4309 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4310 OutputTrack *outputTrack = new OutputTrack(thread, 4311 this, 4312 mSampleRate, 4313 mFormat, 4314 mChannelMask, 4315 frameCount, 4316 IPCThreadState::self()->getCallingUid()); 4317 if (outputTrack->cblk() != NULL) { 4318 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4319 mOutputTracks.add(outputTrack); 4320 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4321 updateWaitTime_l(); 4322 } 4323} 4324 4325void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4326{ 4327 Mutex::Autolock _l(mLock); 4328 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4329 if (mOutputTracks[i]->thread() == thread) { 4330 mOutputTracks[i]->destroy(); 4331 mOutputTracks.removeAt(i); 4332 updateWaitTime_l(); 4333 return; 4334 } 4335 } 4336 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4337} 4338 4339// caller must hold mLock 4340void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4341{ 4342 mWaitTimeMs = UINT_MAX; 4343 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4344 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4345 if (strong != 0) { 4346 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4347 if (waitTimeMs < mWaitTimeMs) { 4348 mWaitTimeMs = waitTimeMs; 4349 } 4350 } 4351 } 4352} 4353 4354 4355bool AudioFlinger::DuplicatingThread::outputsReady( 4356 const SortedVector< sp<OutputTrack> > &outputTracks) 4357{ 4358 for (size_t i = 0; i < outputTracks.size(); i++) { 4359 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4360 if (thread == 0) { 4361 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4362 outputTracks[i].get()); 4363 return false; 4364 } 4365 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4366 // see note at standby() declaration 4367 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4368 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4369 thread.get()); 4370 return false; 4371 } 4372 } 4373 return true; 4374} 4375 4376uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4377{ 4378 return (mWaitTimeMs * 1000) / 2; 4379} 4380 4381void AudioFlinger::DuplicatingThread::cacheParameters_l() 4382{ 4383 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4384 updateWaitTime_l(); 4385 4386 MixerThread::cacheParameters_l(); 4387} 4388 4389// ---------------------------------------------------------------------------- 4390// Record 4391// ---------------------------------------------------------------------------- 4392 4393AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4394 AudioStreamIn *input, 4395 uint32_t sampleRate, 4396 audio_channel_mask_t channelMask, 4397 audio_io_handle_t id, 4398 audio_devices_t outDevice, 4399 audio_devices_t inDevice 4400#ifdef TEE_SINK 4401 , const sp<NBAIO_Sink>& teeSink 4402#endif 4403 ) : 4404 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4405 mInput(input), mActiveTracksGen(0), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4406 // mRsmpInFrames, mRsmpInFramesP2, mRsmpInUnrel, mRsmpInFront, and mRsmpInRear 4407 // are set by readInputParameters() 4408 // mRsmpInIndex LEGACY 4409 mReqChannelCount(popcount(channelMask)), 4410 mReqSampleRate(sampleRate) 4411 // mBytesRead is only meaningful while active, and so is cleared in start() 4412 // (but might be better to also clear here for dump?) 4413#ifdef TEE_SINK 4414 , mTeeSink(teeSink) 4415#endif 4416{ 4417 snprintf(mName, kNameLength, "AudioIn_%X", id); 4418 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4419 4420 readInputParameters(); 4421} 4422 4423 4424AudioFlinger::RecordThread::~RecordThread() 4425{ 4426 mAudioFlinger->unregisterWriter(mNBLogWriter); 4427 delete[] mRsmpInBuffer; 4428 delete mResampler; 4429 delete[] mRsmpOutBuffer; 4430} 4431 4432void AudioFlinger::RecordThread::onFirstRef() 4433{ 4434 run(mName, PRIORITY_URGENT_AUDIO); 4435} 4436 4437bool AudioFlinger::RecordThread::threadLoop() 4438{ 4439 nsecs_t lastWarning = 0; 4440 4441 inputStandBy(); 4442 4443 // used to verify we've read at least once before evaluating how many bytes were read 4444 bool readOnce = false; 4445 4446 // used to request a deferred sleep, to be executed later while mutex is unlocked 4447 bool doSleep = false; 4448 4449reacquire_wakelock: 4450 sp<RecordTrack> activeTrack; 4451 int activeTracksGen; 4452 { 4453 Mutex::Autolock _l(mLock); 4454 size_t size = mActiveTracks.size(); 4455 activeTracksGen = mActiveTracksGen; 4456 if (size > 0) { 4457 // FIXME an arbitrary choice 4458 activeTrack = mActiveTracks[0]; 4459 acquireWakeLock_l(activeTrack->uid()); 4460 if (size > 1) { 4461 SortedVector<int> tmp; 4462 for (size_t i = 0; i < size; i++) { 4463 tmp.add(mActiveTracks[i]->uid()); 4464 } 4465 updateWakeLockUids_l(tmp); 4466 } 4467 } else { 4468 acquireWakeLock_l(-1); 4469 } 4470 } 4471 4472 // start recording 4473 for (;;) { 4474 TrackBase::track_state activeTrackState; 4475 Vector< sp<EffectChain> > effectChains; 4476 4477 // sleep with mutex unlocked 4478 if (doSleep) { 4479 doSleep = false; 4480 usleep(kRecordThreadSleepUs); 4481 } 4482 4483 { // scope for mLock 4484 Mutex::Autolock _l(mLock); 4485 4486 processConfigEvents_l(); 4487 // return value 'reconfig' is currently unused 4488 bool reconfig = checkForNewParameters_l(); 4489 4490 // check exitPending here because checkForNewParameters_l() and 4491 // checkForNewParameters_l() can temporarily release mLock 4492 if (exitPending()) { 4493 break; 4494 } 4495 4496 // if no active track(s), then standby and release wakelock 4497 size_t size = mActiveTracks.size(); 4498 if (size == 0) { 4499 standbyIfNotAlreadyInStandby(); 4500 // exitPending() can't become true here 4501 releaseWakeLock_l(); 4502 ALOGV("RecordThread: loop stopping"); 4503 // go to sleep 4504 mWaitWorkCV.wait(mLock); 4505 ALOGV("RecordThread: loop starting"); 4506 goto reacquire_wakelock; 4507 } 4508 4509 if (mActiveTracksGen != activeTracksGen) { 4510 activeTracksGen = mActiveTracksGen; 4511 SortedVector<int> tmp; 4512 for (size_t i = 0; i < size; i++) { 4513 tmp.add(mActiveTracks[i]->uid()); 4514 } 4515 updateWakeLockUids_l(tmp); 4516 // FIXME an arbitrary choice 4517 activeTrack = mActiveTracks[0]; 4518 } 4519 4520 if (activeTrack->isTerminated()) { 4521 removeTrack_l(activeTrack); 4522 mActiveTracks.remove(activeTrack); 4523 mActiveTracksGen++; 4524 continue; 4525 } 4526 4527 activeTrackState = activeTrack->mState; 4528 switch (activeTrackState) { 4529 case TrackBase::PAUSING: 4530 standbyIfNotAlreadyInStandby(); 4531 mActiveTracks.remove(activeTrack); 4532 mActiveTracksGen++; 4533 mStartStopCond.broadcast(); 4534 doSleep = true; 4535 continue; 4536 4537 case TrackBase::RESUMING: 4538 mStandby = false; 4539 if (mReqChannelCount != activeTrack->channelCount()) { 4540 mActiveTracks.remove(activeTrack); 4541 mActiveTracksGen++; 4542 mStartStopCond.broadcast(); 4543 continue; 4544 } 4545 if (readOnce) { 4546 mStartStopCond.broadcast(); 4547 // record start succeeds only if first read from audio input succeeds 4548 if (mBytesRead < 0) { 4549 mActiveTracks.remove(activeTrack); 4550 mActiveTracksGen++; 4551 continue; 4552 } 4553 activeTrack->mState = TrackBase::ACTIVE; 4554 } 4555 break; 4556 4557 case TrackBase::ACTIVE: 4558 break; 4559 4560 case TrackBase::IDLE: 4561 doSleep = true; 4562 continue; 4563 4564 default: 4565 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4566 } 4567 4568 lockEffectChains_l(effectChains); 4569 } 4570 4571 // thread mutex is now unlocked, mActiveTracks unknown, activeTrack != 0, kept, immutable 4572 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4573 4574 for (size_t i = 0; i < effectChains.size(); i ++) { 4575 // thread mutex is not locked, but effect chain is locked 4576 effectChains[i]->process_l(); 4577 } 4578 4579 AudioBufferProvider::Buffer buffer; 4580 buffer.frameCount = mFrameCount; 4581 status_t status = activeTrack->getNextBuffer(&buffer); 4582 if (status == NO_ERROR) { 4583 readOnce = true; 4584 size_t framesOut = buffer.frameCount; 4585 if (mResampler == NULL) { 4586 // no resampling 4587 while (framesOut) { 4588 size_t framesIn = mFrameCount - mRsmpInIndex; 4589 if (framesIn > 0) { 4590 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4591 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4592 activeTrack->mFrameSize; 4593 if (framesIn > framesOut) { 4594 framesIn = framesOut; 4595 } 4596 mRsmpInIndex += framesIn; 4597 framesOut -= framesIn; 4598 if (mChannelCount == mReqChannelCount) { 4599 memcpy(dst, src, framesIn * mFrameSize); 4600 } else { 4601 if (mChannelCount == 1) { 4602 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4603 (int16_t *)src, framesIn); 4604 } else { 4605 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4606 (int16_t *)src, framesIn); 4607 } 4608 } 4609 } 4610 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4611 void *readInto; 4612 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4613 readInto = buffer.raw; 4614 framesOut = 0; 4615 } else { 4616 readInto = mRsmpInBuffer; 4617 mRsmpInIndex = 0; 4618 } 4619 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4620 mBufferSize); 4621 if (mBytesRead <= 0) { 4622 // TODO: verify that it's benign to use a stale track state 4623 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4624 { 4625 ALOGE("Error reading audio input"); 4626 // Force input into standby so that it tries to 4627 // recover at next read attempt 4628 inputStandBy(); 4629 doSleep = true; 4630 } 4631 mRsmpInIndex = mFrameCount; 4632 framesOut = 0; 4633 buffer.frameCount = 0; 4634 } 4635#ifdef TEE_SINK 4636 else if (mTeeSink != 0) { 4637 (void) mTeeSink->write(readInto, 4638 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4639 } 4640#endif 4641 } 4642 } 4643 } else { 4644 // resampling 4645 4646 // avoid busy-waiting if client doesn't keep up 4647 bool madeProgress = false; 4648 4649 // keep mRsmpInBuffer full so resampler always has sufficient input 4650 for (;;) { 4651 int32_t rear = mRsmpInRear; 4652 ssize_t filled = rear - mRsmpInFront; 4653 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 4654 // exit once there is enough data in buffer for resampler 4655 if ((size_t) filled >= mRsmpInFrames) { 4656 break; 4657 } 4658 size_t avail = mRsmpInFramesP2 - filled; 4659 // Only try to read full HAL buffers. 4660 // But if the HAL read returns a partial buffer, use it. 4661 if (avail < mFrameCount) { 4662 ALOGE("insufficient space to read: avail %d < mFrameCount %d", 4663 avail, mFrameCount); 4664 break; 4665 } 4666 // If 'avail' is non-contiguous, first read past the nominal end of buffer, then 4667 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4668 rear &= mRsmpInFramesP2 - 1; 4669 mBytesRead = mInput->stream->read(mInput->stream, 4670 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4671 if (mBytesRead <= 0) { 4672 ALOGE("read failed: mBytesRead=%d < %u", mBytesRead, mBufferSize); 4673 break; 4674 } 4675 ALOG_ASSERT((size_t) mBytesRead <= mBufferSize); 4676 size_t framesRead = mBytesRead / mFrameSize; 4677 ALOG_ASSERT(framesRead > 0); 4678 madeProgress = true; 4679 // If 'avail' was non-contiguous, we now correct for reading past end of buffer. 4680 size_t part1 = mRsmpInFramesP2 - rear; 4681 if (framesRead > part1) { 4682 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4683 (framesRead - part1) * mFrameSize); 4684 } 4685 mRsmpInRear += framesRead; 4686 } 4687 4688 if (!madeProgress) { 4689 ALOGV("Did not make progress"); 4690 usleep(((mFrameCount * 1000) / mSampleRate) * 1000); 4691 } 4692 4693 // resampler accumulates, but we only have one source track 4694 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4695 mResampler->resample(mRsmpOutBuffer, framesOut, 4696 this /* AudioBufferProvider* */); 4697 // ditherAndClamp() works as long as all buffers returned by 4698 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4699 if (mReqChannelCount == 1) { 4700 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4701 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4702 // the resampler always outputs stereo samples: 4703 // do post stereo to mono conversion 4704 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4705 framesOut); 4706 } else { 4707 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4708 } 4709 // now done with mRsmpOutBuffer 4710 4711 } 4712 if (mFramestoDrop == 0) { 4713 activeTrack->releaseBuffer(&buffer); 4714 } else { 4715 if (mFramestoDrop > 0) { 4716 mFramestoDrop -= buffer.frameCount; 4717 if (mFramestoDrop <= 0) { 4718 clearSyncStartEvent(); 4719 } 4720 } else { 4721 mFramestoDrop += buffer.frameCount; 4722 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4723 mSyncStartEvent->isCancelled()) { 4724 ALOGW("Synced record %s, session %d, trigger session %d", 4725 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4726 activeTrack->sessionId(), 4727 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4728 clearSyncStartEvent(); 4729 } 4730 } 4731 } 4732 activeTrack->clearOverflow(); 4733 } 4734 // client isn't retrieving buffers fast enough 4735 else { 4736 if (!activeTrack->setOverflow()) { 4737 nsecs_t now = systemTime(); 4738 if ((now - lastWarning) > kWarningThrottleNs) { 4739 ALOGW("RecordThread: buffer overflow"); 4740 lastWarning = now; 4741 } 4742 } 4743 // Release the processor for a while before asking for a new buffer. 4744 // This will give the application more chance to read from the buffer and 4745 // clear the overflow. 4746 doSleep = true; 4747 } 4748 4749 // enable changes in effect chain 4750 unlockEffectChains(effectChains); 4751 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4752 } 4753 4754 standbyIfNotAlreadyInStandby(); 4755 4756 { 4757 Mutex::Autolock _l(mLock); 4758 for (size_t i = 0; i < mTracks.size(); i++) { 4759 sp<RecordTrack> track = mTracks[i]; 4760 track->invalidate(); 4761 } 4762 mActiveTracks.clear(); 4763 mActiveTracksGen++; 4764 mStartStopCond.broadcast(); 4765 } 4766 4767 releaseWakeLock(); 4768 4769 ALOGV("RecordThread %p exiting", this); 4770 return false; 4771} 4772 4773void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 4774{ 4775 if (!mStandby) { 4776 inputStandBy(); 4777 mStandby = true; 4778 } 4779} 4780 4781void AudioFlinger::RecordThread::inputStandBy() 4782{ 4783 mInput->stream->common.standby(&mInput->stream->common); 4784} 4785 4786sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4787 const sp<AudioFlinger::Client>& client, 4788 uint32_t sampleRate, 4789 audio_format_t format, 4790 audio_channel_mask_t channelMask, 4791 size_t *pFrameCount, 4792 int sessionId, 4793 int uid, 4794 IAudioFlinger::track_flags_t *flags, 4795 pid_t tid, 4796 status_t *status) 4797{ 4798 size_t frameCount = *pFrameCount; 4799 sp<RecordTrack> track; 4800 status_t lStatus; 4801 4802 lStatus = initCheck(); 4803 if (lStatus != NO_ERROR) { 4804 ALOGE("createRecordTrack_l() audio driver not initialized"); 4805 goto Exit; 4806 } 4807 // client expresses a preference for FAST, but we get the final say 4808 if (*flags & IAudioFlinger::TRACK_FAST) { 4809 if ( 4810 // use case: callback handler and frame count is default or at least as large as HAL 4811 ( 4812 (tid != -1) && 4813 ((frameCount == 0) || 4814 (frameCount >= mFrameCount)) 4815 ) && 4816 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4817 // mono or stereo 4818 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4819 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4820 // hardware sample rate 4821 (sampleRate == mSampleRate) && 4822 // record thread has an associated fast recorder 4823 hasFastRecorder() 4824 // FIXME test that RecordThread for this fast track has a capable output HAL 4825 // FIXME add a permission test also? 4826 ) { 4827 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4828 if (frameCount == 0) { 4829 frameCount = mFrameCount * kFastTrackMultiplier; 4830 } 4831 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4832 frameCount, mFrameCount); 4833 } else { 4834 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4835 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4836 "hasFastRecorder=%d tid=%d", 4837 frameCount, mFrameCount, format, 4838 audio_is_linear_pcm(format), 4839 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4840 *flags &= ~IAudioFlinger::TRACK_FAST; 4841 // For compatibility with AudioRecord calculation, buffer depth is forced 4842 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4843 // This is probably too conservative, but legacy application code may depend on it. 4844 // If you change this calculation, also review the start threshold which is related. 4845 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4846 size_t mNormalFrameCount = 2048; // FIXME 4847 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4848 if (minBufCount < 2) { 4849 minBufCount = 2; 4850 } 4851 size_t minFrameCount = mNormalFrameCount * minBufCount; 4852 if (frameCount < minFrameCount) { 4853 frameCount = minFrameCount; 4854 } 4855 } 4856 } 4857 *pFrameCount = frameCount; 4858 4859 // FIXME use flags and tid similar to createTrack_l() 4860 4861 { // scope for mLock 4862 Mutex::Autolock _l(mLock); 4863 4864 track = new RecordTrack(this, client, sampleRate, 4865 format, channelMask, frameCount, sessionId, uid); 4866 4867 lStatus = track->initCheck(); 4868 if (lStatus != NO_ERROR) { 4869 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 4870 // track must be cleared from the caller as the caller has the AF lock 4871 goto Exit; 4872 } 4873 mTracks.add(track); 4874 4875 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4876 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4877 mAudioFlinger->btNrecIsOff(); 4878 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4879 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4880 4881 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4882 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4883 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4884 // so ask activity manager to do this on our behalf 4885 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4886 } 4887 } 4888 lStatus = NO_ERROR; 4889 4890Exit: 4891 *status = lStatus; 4892 return track; 4893} 4894 4895status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4896 AudioSystem::sync_event_t event, 4897 int triggerSession) 4898{ 4899 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4900 sp<ThreadBase> strongMe = this; 4901 status_t status = NO_ERROR; 4902 4903 if (event == AudioSystem::SYNC_EVENT_NONE) { 4904 clearSyncStartEvent(); 4905 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4906 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4907 triggerSession, 4908 recordTrack->sessionId(), 4909 syncStartEventCallback, 4910 this); 4911 // Sync event can be cancelled by the trigger session if the track is not in a 4912 // compatible state in which case we start record immediately 4913 if (mSyncStartEvent->isCancelled()) { 4914 clearSyncStartEvent(); 4915 } else { 4916 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4917 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4918 } 4919 } 4920 4921 { 4922 // This section is a rendezvous between binder thread executing start() and RecordThread 4923 AutoMutex lock(mLock); 4924 if (mActiveTracks.size() > 0) { 4925 // FIXME does not work for multiple active tracks 4926 if (mActiveTracks.indexOf(recordTrack) != 0) { 4927 status = -EBUSY; 4928 } else if (recordTrack->mState == TrackBase::PAUSING) { 4929 recordTrack->mState = TrackBase::ACTIVE; 4930 } 4931 return status; 4932 } 4933 4934 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4935 recordTrack->mState = TrackBase::IDLE; 4936 mActiveTracks.add(recordTrack); 4937 mActiveTracksGen++; 4938 mLock.unlock(); 4939 status_t status = AudioSystem::startInput(mId); 4940 mLock.lock(); 4941 // FIXME should verify that mActiveTrack is still == recordTrack 4942 if (status != NO_ERROR) { 4943 mActiveTracks.remove(recordTrack); 4944 mActiveTracksGen++; 4945 clearSyncStartEvent(); 4946 return status; 4947 } 4948 // FIXME LEGACY 4949 mRsmpInIndex = mFrameCount; 4950 mRsmpInFront = 0; 4951 mRsmpInRear = 0; 4952 mRsmpInUnrel = 0; 4953 mBytesRead = 0; 4954 if (mResampler != NULL) { 4955 mResampler->reset(); 4956 } 4957 // FIXME hijacking a playback track state name which was intended for start after pause; 4958 // here 'STARTING_2' would be more accurate 4959 recordTrack->mState = TrackBase::RESUMING; 4960 // signal thread to start 4961 ALOGV("Signal record thread"); 4962 mWaitWorkCV.broadcast(); 4963 // do not wait for mStartStopCond if exiting 4964 if (exitPending()) { 4965 mActiveTracks.remove(recordTrack); 4966 mActiveTracksGen++; 4967 status = INVALID_OPERATION; 4968 goto startError; 4969 } 4970 // FIXME incorrect usage of wait: no explicit predicate or loop 4971 mStartStopCond.wait(mLock); 4972 if (mActiveTracks.indexOf(recordTrack) < 0) { 4973 ALOGV("Record failed to start"); 4974 status = BAD_VALUE; 4975 goto startError; 4976 } 4977 ALOGV("Record started OK"); 4978 return status; 4979 } 4980 4981startError: 4982 AudioSystem::stopInput(mId); 4983 clearSyncStartEvent(); 4984 return status; 4985} 4986 4987void AudioFlinger::RecordThread::clearSyncStartEvent() 4988{ 4989 if (mSyncStartEvent != 0) { 4990 mSyncStartEvent->cancel(); 4991 } 4992 mSyncStartEvent.clear(); 4993 mFramestoDrop = 0; 4994} 4995 4996void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4997{ 4998 sp<SyncEvent> strongEvent = event.promote(); 4999 5000 if (strongEvent != 0) { 5001 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5002 me->handleSyncStartEvent(strongEvent); 5003 } 5004} 5005 5006void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5007{ 5008 if (event == mSyncStartEvent) { 5009 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5010 // from audio HAL 5011 mFramestoDrop = mFrameCount * 2; 5012 } 5013} 5014 5015bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5016 ALOGV("RecordThread::stop"); 5017 AutoMutex _l(mLock); 5018 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5019 return false; 5020 } 5021 // note that threadLoop may still be processing the track at this point [without lock] 5022 recordTrack->mState = TrackBase::PAUSING; 5023 // do not wait for mStartStopCond if exiting 5024 if (exitPending()) { 5025 return true; 5026 } 5027 // FIXME incorrect usage of wait: no explicit predicate or loop 5028 mStartStopCond.wait(mLock); 5029 // if we have been restarted, recordTrack is in mActiveTracks here 5030 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5031 ALOGV("Record stopped OK"); 5032 return true; 5033 } 5034 return false; 5035} 5036 5037bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5038{ 5039 return false; 5040} 5041 5042status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5043{ 5044#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5045 if (!isValidSyncEvent(event)) { 5046 return BAD_VALUE; 5047 } 5048 5049 int eventSession = event->triggerSession(); 5050 status_t ret = NAME_NOT_FOUND; 5051 5052 Mutex::Autolock _l(mLock); 5053 5054 for (size_t i = 0; i < mTracks.size(); i++) { 5055 sp<RecordTrack> track = mTracks[i]; 5056 if (eventSession == track->sessionId()) { 5057 (void) track->setSyncEvent(event); 5058 ret = NO_ERROR; 5059 } 5060 } 5061 return ret; 5062#else 5063 return BAD_VALUE; 5064#endif 5065} 5066 5067// destroyTrack_l() must be called with ThreadBase::mLock held 5068void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5069{ 5070 track->terminate(); 5071 track->mState = TrackBase::STOPPED; 5072 // active tracks are removed by threadLoop() 5073 if (mActiveTracks.indexOf(track) < 0) { 5074 removeTrack_l(track); 5075 } 5076} 5077 5078void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5079{ 5080 mTracks.remove(track); 5081 // need anything related to effects here? 5082} 5083 5084void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5085{ 5086 dumpInternals(fd, args); 5087 dumpTracks(fd, args); 5088 dumpEffectChains(fd, args); 5089} 5090 5091void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5092{ 5093 const size_t SIZE = 256; 5094 char buffer[SIZE]; 5095 String8 result; 5096 5097 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5098 result.append(buffer); 5099 5100 if (mActiveTracks.size() > 0) { 5101 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5102 result.append(buffer); 5103 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 5104 result.append(buffer); 5105 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5106 result.append(buffer); 5107 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 5108 result.append(buffer); 5109 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 5110 result.append(buffer); 5111 } else { 5112 result.append("No active record client\n"); 5113 } 5114 5115 write(fd, result.string(), result.size()); 5116 5117 dumpBase(fd, args); 5118} 5119 5120void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5121{ 5122 const size_t SIZE = 256; 5123 char buffer[SIZE]; 5124 String8 result; 5125 5126 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 5127 result.append(buffer); 5128 RecordTrack::appendDumpHeader(result); 5129 for (size_t i = 0; i < mTracks.size(); ++i) { 5130 sp<RecordTrack> track = mTracks[i]; 5131 if (track != 0) { 5132 track->dump(buffer, SIZE); 5133 result.append(buffer); 5134 } 5135 } 5136 5137 size_t size = mActiveTracks.size(); 5138 if (size > 0) { 5139 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 5140 result.append(buffer); 5141 RecordTrack::appendDumpHeader(result); 5142 for (size_t i = 0; i < size; ++i) { 5143 sp<RecordTrack> track = mActiveTracks[i]; 5144 track->dump(buffer, SIZE); 5145 result.append(buffer); 5146 } 5147 5148 } 5149 write(fd, result.string(), result.size()); 5150} 5151 5152// AudioBufferProvider interface 5153status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5154{ 5155 int32_t rear = mRsmpInRear; 5156 int32_t front = mRsmpInFront; 5157 ssize_t filled = rear - front; 5158 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 5159 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5160 front &= mRsmpInFramesP2 - 1; 5161 size_t part1 = mRsmpInFramesP2 - front; 5162 if (part1 > (size_t) filled) { 5163 part1 = filled; 5164 } 5165 size_t ask = buffer->frameCount; 5166 ALOG_ASSERT(ask > 0); 5167 if (part1 > ask) { 5168 part1 = ask; 5169 } 5170 if (part1 == 0) { 5171 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5172 ALOGE("RecordThread::getNextBuffer() starved"); 5173 buffer->raw = NULL; 5174 buffer->frameCount = 0; 5175 mRsmpInUnrel = 0; 5176 return NOT_ENOUGH_DATA; 5177 } 5178 5179 buffer->raw = mRsmpInBuffer + front * mChannelCount; 5180 buffer->frameCount = part1; 5181 mRsmpInUnrel = part1; 5182 return NO_ERROR; 5183} 5184 5185// AudioBufferProvider interface 5186void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5187{ 5188 size_t stepCount = buffer->frameCount; 5189 if (stepCount == 0) { 5190 return; 5191 } 5192 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 5193 mRsmpInUnrel -= stepCount; 5194 mRsmpInFront += stepCount; 5195 buffer->raw = NULL; 5196 buffer->frameCount = 0; 5197} 5198 5199bool AudioFlinger::RecordThread::checkForNewParameters_l() 5200{ 5201 bool reconfig = false; 5202 5203 while (!mNewParameters.isEmpty()) { 5204 status_t status = NO_ERROR; 5205 String8 keyValuePair = mNewParameters[0]; 5206 AudioParameter param = AudioParameter(keyValuePair); 5207 int value; 5208 audio_format_t reqFormat = mFormat; 5209 uint32_t reqSamplingRate = mReqSampleRate; 5210 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 5211 5212 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5213 reqSamplingRate = value; 5214 reconfig = true; 5215 } 5216 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5217 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5218 status = BAD_VALUE; 5219 } else { 5220 reqFormat = (audio_format_t) value; 5221 reconfig = true; 5222 } 5223 } 5224 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5225 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5226 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5227 status = BAD_VALUE; 5228 } else { 5229 reqChannelMask = mask; 5230 reconfig = true; 5231 } 5232 } 5233 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5234 // do not accept frame count changes if tracks are open as the track buffer 5235 // size depends on frame count and correct behavior would not be guaranteed 5236 // if frame count is changed after track creation 5237 if (mActiveTracks.size() > 0) { 5238 status = INVALID_OPERATION; 5239 } else { 5240 reconfig = true; 5241 } 5242 } 5243 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5244 // forward device change to effects that have requested to be 5245 // aware of attached audio device. 5246 for (size_t i = 0; i < mEffectChains.size(); i++) { 5247 mEffectChains[i]->setDevice_l(value); 5248 } 5249 5250 // store input device and output device but do not forward output device to audio HAL. 5251 // Note that status is ignored by the caller for output device 5252 // (see AudioFlinger::setParameters() 5253 if (audio_is_output_devices(value)) { 5254 mOutDevice = value; 5255 status = BAD_VALUE; 5256 } else { 5257 mInDevice = value; 5258 // disable AEC and NS if the device is a BT SCO headset supporting those 5259 // pre processings 5260 if (mTracks.size() > 0) { 5261 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5262 mAudioFlinger->btNrecIsOff(); 5263 for (size_t i = 0; i < mTracks.size(); i++) { 5264 sp<RecordTrack> track = mTracks[i]; 5265 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5266 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5267 } 5268 } 5269 } 5270 } 5271 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5272 mAudioSource != (audio_source_t)value) { 5273 // forward device change to effects that have requested to be 5274 // aware of attached audio device. 5275 for (size_t i = 0; i < mEffectChains.size(); i++) { 5276 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5277 } 5278 mAudioSource = (audio_source_t)value; 5279 } 5280 5281 if (status == NO_ERROR) { 5282 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5283 keyValuePair.string()); 5284 if (status == INVALID_OPERATION) { 5285 inputStandBy(); 5286 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5287 keyValuePair.string()); 5288 } 5289 if (reconfig) { 5290 if (status == BAD_VALUE && 5291 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5292 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5293 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5294 <= (2 * reqSamplingRate)) && 5295 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5296 <= FCC_2 && 5297 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 5298 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 5299 status = NO_ERROR; 5300 } 5301 if (status == NO_ERROR) { 5302 readInputParameters(); 5303 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5304 } 5305 } 5306 } 5307 5308 mNewParameters.removeAt(0); 5309 5310 mParamStatus = status; 5311 mParamCond.signal(); 5312 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5313 // already timed out waiting for the status and will never signal the condition. 5314 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5315 } 5316 return reconfig; 5317} 5318 5319String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5320{ 5321 Mutex::Autolock _l(mLock); 5322 if (initCheck() != NO_ERROR) { 5323 return String8(); 5324 } 5325 5326 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5327 const String8 out_s8(s); 5328 free(s); 5329 return out_s8; 5330} 5331 5332void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) { 5333 AudioSystem::OutputDescriptor desc; 5334 const void *param2 = NULL; 5335 5336 switch (event) { 5337 case AudioSystem::INPUT_OPENED: 5338 case AudioSystem::INPUT_CONFIG_CHANGED: 5339 desc.channelMask = mChannelMask; 5340 desc.samplingRate = mSampleRate; 5341 desc.format = mFormat; 5342 desc.frameCount = mFrameCount; 5343 desc.latency = 0; 5344 param2 = &desc; 5345 break; 5346 5347 case AudioSystem::INPUT_CLOSED: 5348 default: 5349 break; 5350 } 5351 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5352} 5353 5354void AudioFlinger::RecordThread::readInputParameters() 5355{ 5356 delete[] mRsmpInBuffer; 5357 // mRsmpInBuffer is always assigned a new[] below 5358 delete[] mRsmpOutBuffer; 5359 mRsmpOutBuffer = NULL; 5360 delete mResampler; 5361 mResampler = NULL; 5362 5363 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5364 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5365 mChannelCount = popcount(mChannelMask); 5366 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5367 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5368 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5369 } 5370 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5371 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5372 mFrameCount = mBufferSize / mFrameSize; 5373 // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to 5374 // 1 full output buffer, regardless of the alignment of the available input. 5375 mRsmpInFrames = mFrameCount * 3; 5376 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5377 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5378 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5379 mRsmpInFront = 0; 5380 mRsmpInRear = 0; 5381 mRsmpInUnrel = 0; 5382 5383 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5384 mResampler = AudioResampler::create(16, (int) mChannelCount, mReqSampleRate); 5385 mResampler->setSampleRate(mSampleRate); 5386 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5387 // resampler always outputs stereo 5388 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5389 } 5390 mRsmpInIndex = mFrameCount; 5391} 5392 5393uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5394{ 5395 Mutex::Autolock _l(mLock); 5396 if (initCheck() != NO_ERROR) { 5397 return 0; 5398 } 5399 5400 return mInput->stream->get_input_frames_lost(mInput->stream); 5401} 5402 5403uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5404{ 5405 Mutex::Autolock _l(mLock); 5406 uint32_t result = 0; 5407 if (getEffectChain_l(sessionId) != 0) { 5408 result = EFFECT_SESSION; 5409 } 5410 5411 for (size_t i = 0; i < mTracks.size(); ++i) { 5412 if (sessionId == mTracks[i]->sessionId()) { 5413 result |= TRACK_SESSION; 5414 break; 5415 } 5416 } 5417 5418 return result; 5419} 5420 5421KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5422{ 5423 KeyedVector<int, bool> ids; 5424 Mutex::Autolock _l(mLock); 5425 for (size_t j = 0; j < mTracks.size(); ++j) { 5426 sp<RecordThread::RecordTrack> track = mTracks[j]; 5427 int sessionId = track->sessionId(); 5428 if (ids.indexOfKey(sessionId) < 0) { 5429 ids.add(sessionId, true); 5430 } 5431 } 5432 return ids; 5433} 5434 5435AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5436{ 5437 Mutex::Autolock _l(mLock); 5438 AudioStreamIn *input = mInput; 5439 mInput = NULL; 5440 return input; 5441} 5442 5443// this method must always be called either with ThreadBase mLock held or inside the thread loop 5444audio_stream_t* AudioFlinger::RecordThread::stream() const 5445{ 5446 if (mInput == NULL) { 5447 return NULL; 5448 } 5449 return &mInput->stream->common; 5450} 5451 5452status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5453{ 5454 // only one chain per input thread 5455 if (mEffectChains.size() != 0) { 5456 return INVALID_OPERATION; 5457 } 5458 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5459 5460 chain->setInBuffer(NULL); 5461 chain->setOutBuffer(NULL); 5462 5463 checkSuspendOnAddEffectChain_l(chain); 5464 5465 mEffectChains.add(chain); 5466 5467 return NO_ERROR; 5468} 5469 5470size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5471{ 5472 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5473 ALOGW_IF(mEffectChains.size() != 1, 5474 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5475 chain.get(), mEffectChains.size(), this); 5476 if (mEffectChains.size() == 1) { 5477 mEffectChains.removeAt(0); 5478 } 5479 return 0; 5480} 5481 5482}; // namespace android 5483